From loi.dangthanh at gmail.com Thu Jun 1 02:27:26 2017 From: loi.dangthanh at gmail.com (=?UTF-8?B?TOG7o2kgxJDhurduZw==?=) Date: Thu, 01 Jun 2017 02:27:26 +0000 Subject: [Freeswitch-users] testing 1234 In-Reply-To: <1b0c01d2da5d$011a7310$034f5930$@freeswitch.org> References: <1aa001d2da4b$ab19c1c0$014d4540$@freeswitch.org> <1abd01d2da54$c34b84a0$49e28de0$@freeswitch.org> <592F04DE020000310000A092@mail.tedssupply.com> <1b0c01d2da5d$011a7310$034f5930$@freeswitch.org> Message-ID: what's up guys? FS mailing is dead for several days? I didn't receive any mail until this ping-pong today. rgds, Loi Dang On Thu, Jun 1, 2017 at 5:27 AM Ken Rice wrote: > Thanks for replying everyone. > > > > And just a reminder for those that are still on the list and want to > unsubscribe, or just want to manage their list membership, there is a link > at the bottom of every email that comes across the list that you can click. > It will allow you to unsubscribe, or even change you from individual emails > on the list to the digest where you get 1 email a day with all the other > emails aggregated up. > > > > Thanks Guys! > > Ken > > > > > > > > *From:* FreeSWITCH-users [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Matt Broad > *Sent:* Wednesday, May 31, 2017 5:20 PM > > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] testing 1234 > > > > roger roger > > > Matt Broad > > Tel: +44 (0)203 011 1313 <+44%2020%203011%201313> > > Web: www.supportedbusiness.com > > > > On 31 May 2017 at 23:15, John Dalrymple wrote: > > unsubscribe (please) > > > John Dalrymple > > President > > 813-990-0996 <(813)%20990-0996> call or text > > > > > > > > > > On Wed, May 31, 2017 at 6:01 PM, admin wrote: > > Loud and clear! > > > > >>> "Ken Rice" 05/31/17 5:33 PM >>> > > You got this? > > > > *From:* FreeSWITCH-users [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Brian West > *Sent:* Wednesday, May 31, 2017 3:26 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] testing 1234 > > > > I got this. > > > > /b > > > > > > On Wed, May 31, 2017 at 3:22 PM, Ken Rice wrote: > > This is just a test > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > -- > > *Brian West* > brian at freeswitch.org > > *Twitter: @FreeSWITCH , @briankwest* > > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > Book a phone call (CST) > > Allison prompts for FreeSWITCH: > > *https://www.gofundme.com/allison-prompts-for-freeswitch* > > > Got Bugs? Report them here ! | Reddit: > /r/freeswitch > > *T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002 <(918)%20420-9002> > | *M:*+1918424WEST (9378) > *Skype:*briankwest > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian at freeswitch.org Thu Jun 1 03:19:36 2017 From: brian at freeswitch.org (Brian West) Date: Thu, 01 Jun 2017 03:19:36 +0000 Subject: [Freeswitch-users] testing 1234 In-Reply-To: References: <1aa001d2da4b$ab19c1c0$014d4540$@freeswitch.org> <1abd01d2da54$c34b84a0$49e28de0$@freeswitch.org> <592F04DE020000310000A092@mail.tedssupply.com> <1b0c01d2da5d$011a7310$034f5930$@freeswitch.org> Message-ID: We are doing upgrades, a small snafu caused this! Hopefully we can prevent these types of human errors in the future. /b On Wed, May 31, 2017 at 9:30 PM Lợi Đặng wrote: > what's up guys? FS mailing is dead for several days? > I didn't receive any mail until this ping-pong today. > rgds, > Loi Dang > > On Thu, Jun 1, 2017 at 5:27 AM Ken Rice wrote: > >> Thanks for replying everyone. >> >> >> >> And just a reminder for those that are still on the list and want to >> unsubscribe, or just want to manage their list membership, there is a link >> at the bottom of every email that comes across the list that you can click. >> It will allow you to unsubscribe, or even change you from individual emails >> on the list to the digest where you get 1 email a day with all the other >> emails aggregated up. >> >> >> >> Thanks Guys! >> >> Ken >> >> >> >> >> >> >> >> *From:* FreeSWITCH-users [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Matt Broad >> *Sent:* Wednesday, May 31, 2017 5:20 PM >> >> >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] testing 1234 >> >> >> >> roger roger >> >> >> Matt Broad >> >> Tel: +44 (0)203 011 1313 <+44%2020%203011%201313> >> >> Web: www.supportedbusiness.com >> >> >> >> On 31 May 2017 at 23:15, John Dalrymple wrote: >> >> unsubscribe (please) >> >> >> John Dalrymple >> >> President >> >> 813-990-0996 <(813)%20990-0996> call or text >> >> >> >> >> >> >> >> >> >> On Wed, May 31, 2017 at 6:01 PM, admin wrote: >> >> Loud and clear! >> >> >> >> >>> "Ken Rice" 05/31/17 5:33 PM >>> >> >> You got this? >> >> >> >> *From:* FreeSWITCH-users [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Brian West >> *Sent:* Wednesday, May 31, 2017 3:26 PM >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] testing 1234 >> >> >> >> I got this. >> >> >> >> /b >> >> >> >> >> >> On Wed, May 31, 2017 at 3:22 PM, Ken Rice wrote: >> >> This is just a test >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> >> -- >> >> *Brian West* >> brian at freeswitch.org >> >> *Twitter: @FreeSWITCH , @briankwest* >> >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> Book a phone call (CST) >> >> Allison prompts for FreeSWITCH: >> >> *https://www.gofundme.com/allison-prompts-for-freeswitch* >> >> >> Got Bugs? Report them here ! | Reddit: >> /r/freeswitch >> >> *T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002 <(918)%20420-9002> >> | *M:*+1918424WEST (9378) >> *Skype:*briankwest >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at 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 at 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 at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Book a phone call (CST) Allison prompts for FreeSWITCH: *https://www.gofundme.com/allison-prompts-for-freeswitch* Got Bugs? Report them here ! | Reddit: /r/freeswitch *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: From fvillarroel at yahoo.com Thu Jun 1 03:51:31 2017 From: fvillarroel at yahoo.com (FERNANDO VILLARROEL) Date: Thu, 1 Jun 2017 03:51:31 +0000 (UTC) Subject: [Freeswitch-users] FS 1.6 Video References: <990207077.4430.1496289091672.ref@mail.yahoo.com> Message-ID: <990207077.4430.1496289091672@mail.yahoo.com> Dear All. I am follow https://freeswitch.org/confluence/display/FREESWITCH/Debian+8+Jessie But not works for me i received: root at fswebrtc:/etc/apt# apt-get install -y --force-yes freeswitch-video-deps-mostLeyendo lista de paquetes... HechoCreando árbol de dependenciasLeyendo la información de estado... HechoNo se pudieron instalar algunos paquetes. Esto puede significar queusted pidió una situación imposible o, si está usando la distribucióninestable, que algunos paquetes necesarios aún no se han creado o sehan sacado de «Incoming».La siguiente información puede ayudar a resolver la situación: Los siguientes paquetes tienen dependencias incumplidas: freeswitch-video-deps-most : Depende: libavcodec-extra pero no va a instalarseE: No se pudieron corregir los problemas, usted ha retenido paquetes rotos. My /etc/apt/sources.list.d/freeswitch.list deb http://files.freeswitch.org/repo/deb/debian/ jessie maindeb http://files.freeswitch.org/repo/deb/debian-unstable/ jessie main uname -aLinux fswebrtc 3.16.0-4-amd64 #1 SMP Debian 3.16.7-ckt20-1+deb8u3 (2016-01-17) x86_64 GNU/Linux root at fswebrtc:/etc/apt# grep -r deb /etc/apt/ /etc/apt/apt.conf.d/70debconf:// Pre-configure all packages with debconf before they are installed./etc/apt/apt.conf.d/50unattended-upgrades://     site          (eg, "http.debian.net")/etc/apt/apt.conf.d/50unattended-upgrades:// "apt-cache policy", and can be debugged by running/etc/apt/apt.conf.d/50unattended-upgrades:// derived from /etc/debian_version:/etc/apt/sources.list:# deb cdrom:[Debian GNU/Linux 8.3.0 _Jessie_ - Official amd64 NETINST Binary-1 20160123-18:59]/ jessie main/etc/apt/sources.list:#deb cdrom:[Debian GNU/Linux 8.3.0 _Jessie_ - Official amd64 NETINST Binary-1 20160123-18:59]/ jessie main/etc/apt/sources.list:deb http://http.debian.net/debian jessie main/etc/apt/sources.list:deb-src http://http.debian.net/debian jessie main/etc/apt/sources.list:deb http://security.debian.org/ jessie/updates main/etc/apt/sources.list:deb-src http://security.debian.org/ jessie/updates main/etc/apt/sources.list:deb http://http.debian.net/debian jessie-updates main/etc/apt/sources.list:deb-src http://http.debian.net/debian jessie-updates main/etc/apt/sources.list:deb http://www.deb-multimedia.org jessie main non-freeCoincidencia en el fichero binario /etc/apt/trusted.gpg.d/debian-archive-jessie-stable.gpgCoincidencia en el fichero binario /etc/apt/trusted.gpg.d/debian-archive-squeeze-stable.gpgCoincidencia en el fichero binario /etc/apt/trusted.gpg.d/debian-archive-jessie-security-automatic.gpgCoincidencia en el fichero binario /etc/apt/trusted.gpg.d/debian-archive-squeeze-automatic.gpgCoincidencia en el fichero binario /etc/apt/trusted.gpg.d/deb-multimedia-keyring.gpgCoincidencia en el fichero binario /etc/apt/trusted.gpg.d/debian-archive-wheezy-stable.gpgCoincidencia en el fichero binario /etc/apt/trusted.gpg.d/debian-archive-jessie-automatic.gpgCoincidencia en el fichero binario /etc/apt/trusted.gpg.d/debian-archive-wheezy-automatic.gpg/etc/apt/sources.list.d/freeswitch.list:deb http://files.freeswitch.org/repo/deb/debian/ jessie main/etc/apt/sources.list.d/freeswitch.list:deb http://files.freeswitch.org/repo/deb/debian-unstable/ jessie main I appreciate some help or tips please. -------------- next part -------------- An HTML attachment was scrubbed... URL: From tculjaga at gmail.com Thu Jun 1 06:46:33 2017 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Thu, 1 Jun 2017 08:46:33 +0200 Subject: [Freeswitch-users] originated_legs In-Reply-To: References: Message-ID: anyone ? :=) On 30 May 2017 at 14:40, Tihomir Culjaga wrote: > Hello, > > There is something that is bugging me pretty hard and i need to understand > how variable_originated_legs in CHANNEL_HANGUP gets populated ? > > > example: > > variable_originated_legs: ARRAY::faf71922-4526-11e7-b0f4-9fb65fdc7341;Outbound > Call;888|:faf71922-4526-11e7-b0f4-9fb65fdc7341;Outbound > Call;888|:faf76c56-4526-11e7-b0fd-9fb65fdc7341;Outbound Call;0916331550", > > > > > the reason :=) > > i have a scenario like this: > > > 1. Incoming calls (IN_CALL) are sent to PARK application > 2. On CHANNEL_PARK i originate a call to an extension say EXT1 > > e.g. originate {ogirination_uuid=}user/1002 & park() ... and > eventually i do uuid_bridge > > > 3. EXT1 makes a call transfer to EXT2 (using att_xfer) and hangs up the > call > 4. on CHANNEL_HANGUP from EXT1 i learn EXT2 uuid and.,,, IN_CALL is > talking to EXT2 and we are happy :=) > > > ... this is all good if the transfer destination is a single extension (i > get enough info from signal_bond etc...). But if i have multiple > destination behind EXT2 e.g. user/1002,user/1003,sofia/gateway/gw1/012345 > the thing gets complicated. > > I was hoping i could exploit variable_originated_leg in CHANNEL_HANGUP > from EXT1 to learn all origination legs uuid after an att_xfer to a > multiple destination. > > is that feasible ? I mean, will this variable contain all UUIDs att_xfer > generated on a multi-destination transfer ? > > > > please i would appreciate any heads up here :=) > > thanks, > Tihomir. > -------------- next part -------------- An HTML attachment was scrubbed... URL: From alexdruzhilov at gmail.com Thu Jun 1 09:08:27 2017 From: alexdruzhilov at gmail.com (=?UTF-8?B?0JDQu9C10LrRgdCw0L3QtNGAINCU0YDRg9C20LjQu9C+0LI=?=) Date: Thu, 1 Jun 2017 12:08:27 +0300 Subject: [Freeswitch-users] Video conference with transcode mode is not working more than for 5-6 users Message-ID: OS: CentOS 7.2.1511 Freeswitch: 1.6.17 Steps: 1) create video conference (mod_sofia and conference-mode = transcode) with one video stream from conference owner 2) add 4-6 members in this conference who will receive owner's video stream 3) everyone in this conference will see dramatic degradation of video stream (incoming video stream bitrate falls from 1 Mbps to 0.5 Mbps, frame rate falls down to 10 fps from 30 fps, lag between video and audio stream appears) But I don't see any problems with CPU, memory or network. So does anybody knows whether it is an issue or it's how freeswitch 'transcode' mode should works? And what to do to tune performance of this mode? -------------- next part -------------- An HTML attachment was scrubbed... URL: From dist.lists at gmail.com Thu Jun 1 10:55:55 2017 From: dist.lists at gmail.com (Dist Lists) Date: Thu, 1 Jun 2017 12:55:55 +0200 Subject: [Freeswitch-users] replace b-leg in a bridge on 302 moved temporarily Message-ID: Hi, I am using ESL for my dial plan and I have the following scenario. Leg-a is being bridged with the bridge() app to multiple endpoints (b-legs). One of the b-legs sends 302 Moved Temporarily back. What I'd like to do is to process the new destination number from the 302, because it may be mapped to multiple endpoints and after figuring out the additional b-leg endpoints to initiate calls to them, while at same time also keeping the original (already ringing) b-legs from the initial call. Obviously on answer the wining b-leg needs to be bridged to the a-leg, regardless if it is an "original" b-leg or a "new" one. To put it another way, I'd like to somehow "replace" the b-leg that sent the 302 Moved Temporarily with multiple new endpoints (b-legs), while also keeping the rest of the originally called b-legs. Sort of adding additional endpoints via ESL to the bridge() application after the fact. And the whole setup needs to work with bypass_media. There is more to the story like the need to check if the endpoint is authorized to redirect the call to the new destination and so on, but I'm skipping this for the sake of simplicity. I have been trying to find a solution or a workaround for a few days now, but nothing worked. What I tested so far: - letting Freeswith handle the 302 within the stack doesn't work, because I cannot handle the mapping of the destination number to multiple endpoints. - manual-redirect, which transfers the a-leg in the redirect context, but also cancels all the other b-legs, didn't help either. - outbound_redirect_fatal=true leaves the other b-legs ringing, but I cannot "add" new b-legs to the bridge() app. - originate-ing the b-legs via API and then trying to somehow link them within the dial plan to the a-leg. This kind of works if I use api_on_ring, api_on_media, etc to pass progress to the a-leg, but it feels too fragile. Most importantly I couldn't figure out how to make bypass_media work, since the a-leg and the b-legs are only loosely linked to one another via the dial plan, but Freeswitch itself doesn't know anything about this relation. I even tied setting some variables like originate_signal_bond and originator, but it didn't help either. - loopback endpoints, which I tried hoping that the loopback a-leg, which can be transferred in a new context after a 302, will allow me to call multiple new endpoint. Unfortunately this doesn't seem to work with bypass_media. I am out of ideas what to try next and was hoping for some feedback from the list. It looks like what I'm trying to achieve is not possible with the current set of features. If that is indeed the case, what would be the best approach to add this functionality? Maybe by extending an existing application/api? -------------- next part -------------- An HTML attachment was scrubbed... URL: From Paul.Mateer at outlook.com Thu Jun 1 12:20:34 2017 From: Paul.Mateer at outlook.com (Paul Mateer) Date: Thu, 1 Jun 2017 12:20:34 +0000 Subject: [Freeswitch-users] Embedding FreeSWITCH Message-ID: Guys, I'm looking to embed FreeSWITCH into a piece of Windows software which is extendable through the use of add-ins (written in C#). When installed the software has an add-ins folder containing subfolders for each add-in to be loaded. By default FreeSWITCH would expect the base directory to be that of the executable, but in this case I want it to be down in the add-ins subfolder since it makes more sense to have it with the add-in rather than the application. I figured that this could be solved easily enough by adding the following line of code before the call to switch_core_init(): freeswitch.switch_core_init((uint)flags, switch_bool_t.SWITCH_FALSE, out err); However doing this seems to cause a problem when trying to load certain modules (in particular mod_sofia.dll). This seems to be because the new base directory that I have specified is not included in the alternate search path (set using SetDefaultDllDirectories and AddDllDirectory) so when a call is made to switch_dso_open to load the module it doesn't search that directory for modules such as pthread.dll. Now I could call AddDllDirectory as part of my freeSWITCH initialization code and then call RemoveDllDirectory when shutting down, but I was wondering if there was a possible justification for this being done in the CSharp_switch_directories_base_dir_set method of freeswitch_wrap.cxx? Interestingly there is also a bug in the switch_dso_open method as the second call to LoadLibraryEx doesn't capture the return value in the lib variable, so even if the second call to LoadLibraryEx succeeds switch_dso_open will return NULL. Paul -------------- next part -------------- An HTML attachment was scrubbed... URL: From rick at magicmail.mooo.com Thu Jun 1 12:20:41 2017 From: rick at magicmail.mooo.com (Rick Jarvis) Date: Thu, 1 Jun 2017 13:20:41 +0100 Subject: [Freeswitch-users] FS to FS In-Reply-To: <5BD3EEFB-4F3C-4E14-B80C-1CED60244C5E@magicmail.mooo.com> References: <5BD3EEFB-4F3C-4E14-B80C-1CED60244C5E@magicmail.mooo.com> Message-ID: Apologies if you’ve seen this before, but not sure if anyone got it due to the mail issues and I’m still scratching my hairless head ;) ... > On 31 May 2017, at 14:57, Rick Jarvis wrote: > > Sorry if this is a dumb question, but it’s something I’ve always struggled with that I’m convinced should be quite straightforward. > > Getting one FS to register with another as an extension (I know there are other ways of getting FS to talk together, but this is the situation I’m faced with). > > Box one (the host) is not behind NAT. > Box two (the client) IS behind NAT. > > In my mind it should be good enough to have a directory entry on box one, and a SIP profile on box two, then something like a catchall dial plan entry on box 2 to accept the calls made to the ‘extension’ it’s registered as and handle the call accordingly. But I don’t think it’s that simple? > > I’ve looked through the soft phone configuration, which suggests using a directory entry with the register flag set to true, but this doesn’t seem to help me. > > Thanks > R From Alexander.Haugg at c4b.de Thu Jun 1 13:21:57 2017 From: Alexander.Haugg at c4b.de (Alexander Haugg) Date: Thu, 1 Jun 2017 13:21:57 +0000 Subject: [Freeswitch-users] I get no Emails from the user list! Message-ID: <42ddf3d866e348108134d659dfb04e8b@c4b.de> Hi, for serveral days i get no emails from the user list. My account settings are ok. My local system is checked. What could be the problem? Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: From Alexander.Haugg at c4b.de Thu Jun 1 13:24:20 2017 From: Alexander.Haugg at c4b.de (Alexander Haugg) Date: Thu, 1 Jun 2017 13:24:20 +0000 Subject: [Freeswitch-users] WG: Turn configuration mod_sofia Message-ID: Hi Brian, we use the Freeswitch as "Man in the Middle" for WebRTC. The WebRTC Clients (ICE Link Stack from Frozen Mountain) registered via SIP on the Freeswitch. The SIP signalling is a special szenario and works successfully in several LAN WAN setups. But now we need srflx (that's fine with teh STUN configuration) an relay candidates in the SDP that's ganerated by the Freeswitch. My think is, for this case i need the possibility to configure the turn server with credentials in the sip profile like the stun configuration too. - STUN configuration "" - ICE configuration could be? "" Thanks a lot >From brian at freeswitch.org Tue May 30 13:50:32 2017 From: brian at freeswitch.org (Brian West) Date: Tue, 30 May 2017 08:50:32 -0500 Subject: [Freeswitch-users] Turn configuration mod_sofia In-Reply-To: > References: > Message-ID: > There is little reason to use TURN when speaking to FreeSWITCH, What issue are you trying to solve? Von: Alexander Haugg Gesendet: Montag, 29. Mai 2017 18:25 An: 'freeswitch-users at lists.freeswitch.org' > Betreff: Turn configuration mod_sofia Hi all, the stun configuration for the mod_sofia profile is very easy "" but what is to do, if i need a relay candidate in the sdp? How can i set the turn address and the login credentials? Thanks a lot -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian at freeswitch.org Thu Jun 1 14:02:28 2017 From: brian at freeswitch.org (Brian West) Date: Thu, 1 Jun 2017 09:02:28 -0500 Subject: [Freeswitch-users] FS to FS In-Reply-To: References: <5BD3EEFB-4F3C-4E14-B80C-1CED60244C5E@magicmail.mooo.com> Message-ID: Its no different than setting up a gateway to an ITSP, you'll setup a user in the directory on FS-A, then have FS-B register to it. You could also skip that completely and do an ACL list to authenticate the two systems against each other. /b On Thu, Jun 1, 2017 at 7:20 AM, Rick Jarvis wrote: > Apologies if you’ve seen this before, but not sure if anyone got it due to > the mail issues and I’m still scratching my hairless head ;) ... > > > On 31 May 2017, at 14:57, Rick Jarvis wrote: > > > > Sorry if this is a dumb question, but it’s something I’ve always > struggled with that I’m convinced should be quite straightforward. > > > > Getting one FS to register with another as an extension (I know there > are other ways of getting FS to talk together, but this is the situation > I’m faced with). > > > > Box one (the host) is not behind NAT. > > Box two (the client) IS behind NAT. > > > > In my mind it should be good enough to have a directory entry on box > one, and a SIP profile on box two, then something like a catchall dial plan > entry on box 2 to accept the calls made to the ‘extension’ it’s registered > as and handle the call accordingly. But I don’t think it’s that simple? > > > > I’ve looked through the soft phone configuration, which suggests using a > directory entry with the register flag set to true, but this doesn’t seem > to help me. > > > > Thanks > > R > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Book a phone call (CST) Allison prompts for FreeSWITCH: *https://www.gofundme.com/allison-prompts-for-freeswitch* Got Bugs? Report them here ! | Reddit: /r/freeswitch *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian at freeswitch.org Thu Jun 1 17:04:22 2017 From: brian at freeswitch.org (Brian West) Date: Thu, 1 Jun 2017 12:04:22 -0500 Subject: [Freeswitch-users] I get no Emails from the user list! In-Reply-To: <42ddf3d866e348108134d659dfb04e8b@c4b.de> References: <42ddf3d866e348108134d659dfb04e8b@c4b.de> Message-ID: Root cause, Migration of service, localhost was ::1 in /etc/hosts, ::1 wasn't on the relay list, but mail to anyone @freeswitch.org still received email posts, so I failed to notice a problem. /b On Thu, Jun 1, 2017 at 8:21 AM, Alexander Haugg wrote: > Hi, > > > > for serveral days i get no emails from the user list. > > My account settings are ok. > > > > My local system is checked. > > > > What could be the problem? > > > > Thanks! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Book a phone call (CST) Allison prompts for FreeSWITCH: *https://www.gofundme.com/allison-prompts-for-freeswitch* Got Bugs? Report them here ! | Reddit: /r/freeswitch *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: From gmaruzz at gmail.com Thu Jun 1 17:14:05 2017 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Thu, 1 Jun 2017 19:14:05 +0200 Subject: [Freeswitch-users] FS 1.6 Video In-Reply-To: <990207077.4430.1496289091672@mail.yahoo.com> References: <990207077.4430.1496289091672.ref@mail.yahoo.com> <990207077.4430.1496289091672@mail.yahoo.com> Message-ID: Fernando, Be sure to follow the steps exactly. Its important you "apt-get update" after adding the repository, and before i stalling the packages. Hope this helps -giovanni sent from mobile cell: +39 347 266 56 18 Giovanni Maruzzelli OpenTelecom.IT On Jun 1, 2017 5:56 AM, "FERNANDO VILLARROEL" wrote: Dear All. I am follow https://freeswitch.org/confluence/display/FREESWITCH/ Debian+8+Jessie But not works for me i received: root at fswebrtc:/etc/apt# apt-get install -y --force-yes freeswitch-video-deps-most Leyendo lista de paquetes... Hecho Creando árbol de dependencias Leyendo la información de estado... Hecho No se pudieron instalar algunos paquetes. Esto puede significar que usted pidió una situación imposible o, si está usando la distribución inestable, que algunos paquetes necesarios aún no se han creado o se han sacado de «Incoming». La siguiente información puede ayudar a resolver la situación: Los siguientes paquetes tienen dependencias incumplidas: freeswitch-video-deps-most : Depende: libavcodec-extra pero no va a instalarse E: No se pudieron corregir los problemas, usted ha retenido paquetes rotos. My /etc/apt/sources.list.d/freeswitch.list deb http://files.freeswitch.org/repo/deb/debian/ jessie main deb http://files.freeswitch.org/repo/deb/debian-unstable/ jessie main uname -a Linux fswebrtc 3.16.0-4-amd64 #1 SMP Debian 3.16.7-ckt20-1+deb8u3 (2016-01-17) x86_64 GNU/Linux root at fswebrtc:/etc/apt# grep -r deb /etc/apt/ /etc/apt/apt.conf.d/70debconf:// Pre-configure all packages with debconf before they are installed. /etc/apt/apt.conf.d/50unattended-upgrades:// site (eg, " http.debian.net") /etc/apt/apt.conf.d/50unattended-upgrades:// "apt-cache policy", and can be debugged by running /etc/apt/apt.conf.d/50unattended-upgrades:// derived from /etc/debian_version: /etc/apt/sources.list:# deb cdrom:[Debian GNU/Linux 8.3.0 _Jessie_ - Official amd64 NETINST Binary-1 20160123-18:59]/ jessie main /etc/apt/sources.list:#deb cdrom:[Debian GNU/Linux 8.3.0 _Jessie_ - Official amd64 NETINST Binary-1 20160123-18:59]/ jessie main /etc/apt/sources.list:deb http://http.debian.net/debian jessie main /etc/apt/sources.list:deb-src http://http.debian.net/debian jessie main /etc/apt/sources.list:deb http://security.debian.org/ jessie/updates main /etc/apt/sources.list:deb-src http://security.debian.org/ jessie/updates main /etc/apt/sources.list:deb http://http.debian.net/debian jessie-updates main /etc/apt/sources.list:deb-src http://http.debian.net/debian jessie-updates main /etc/apt/sources.list:deb http://www.deb-multimedia.org jessie main non-free Coincidencia en el fichero binario /etc/apt/trusted.gpg.d/debian- archive-jessie-stable.gpg Coincidencia en el fichero binario /etc/apt/trusted.gpg.d/debian- archive-squeeze-stable.gpg Coincidencia en el fichero binario /etc/apt/trusted.gpg.d/debian- archive-jessie-security-automatic.gpg Coincidencia en el fichero binario /etc/apt/trusted.gpg.d/debian- archive-squeeze-automatic.gpg Coincidencia en el fichero binario /etc/apt/trusted.gpg.d/deb- multimedia-keyring.gpg Coincidencia en el fichero binario /etc/apt/trusted.gpg.d/debian- archive-wheezy-stable.gpg Coincidencia en el fichero binario /etc/apt/trusted.gpg.d/debian- archive-jessie-automatic.gpg Coincidencia en el fichero binario /etc/apt/trusted.gpg.d/debian- archive-wheezy-automatic.gpg /etc/apt/sources.list.d/freeswitch.list:deb http://files.freeswitch.org/ repo/deb/debian/ jessie main /etc/apt/sources.list.d/freeswitch.list:deb http://files.freeswitch.org/ repo/deb/debian-unstable/ jessie main I appreciate some help or tips please. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From tculjaga at gmail.com Fri Jun 2 05:23:04 2017 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Fri, 2 Jun 2017 07:23:04 +0200 Subject: [Freeswitch-users] testing 1234 In-Reply-To: References: <1aa001d2da4b$ab19c1c0$014d4540$@freeswitch.org> <1abd01d2da54$c34b84a0$49e28de0$@freeswitch.org> <592F04DE020000310000A092@mail.tedssupply.com> <1b0c01d2da5d$011a7310$034f5930$@freeswitch.org> Message-ID: <00C4FE9F-691C-4A29-9375-9BCE4CB172FB@gmail.com> Still, not a single new email in the list... quite strange ;) Sent from my iPhone > On 1 Jun 2017, at 05:19, Brian West wrote: > > We are doing upgrades, a small snafu caused this! > > Hopefully we can prevent these types of human errors in the future. > > /b > > >> On Wed, May 31, 2017 at 9:30 PM Lợi Đặng wrote: >> what's up guys? FS mailing is dead for several days? >> I didn't receive any mail until this ping-pong today. >> rgds, >> Loi Dang >> >>> On Thu, Jun 1, 2017 at 5:27 AM Ken Rice wrote: >>> Thanks for replying everyone. >>> >>> >>> >>> And just a reminder for those that are still on the list and want to unsubscribe, or just want to manage their list membership, there is a link at the bottom of every email that comes across the list that you can click. It will allow you to unsubscribe, or even change you from individual emails on the list to the digest where you get 1 email a day with all the other emails aggregated up. >>> >>> >>> >>> Thanks Guys! >>> >>> Ken >>> >>> >>> >>> >>> >>> >>> >>> From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Matt Broad >>> Sent: Wednesday, May 31, 2017 5:20 PM >>> >>> >>> To: FreeSWITCH Users Help >>> Subject: Re: [Freeswitch-users] testing 1234 >>> >>> >>> >>> roger roger >>> >>> >>> >>> Matt Broad >>> >>> Tel: +44 (0)203 011 1313 >>> >>> Web: www.supportedbusiness.com >>> >>> >>> >>> >>> >>> On 31 May 2017 at 23:15, John Dalrymple wrote: >>> >>> unsubscribe (please) >>> >>> >>> >>> John Dalrymple >>> >>> President >>> >>> 813-990-0996 call or text >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> On Wed, May 31, 2017 at 6:01 PM, admin wrote: >>> >>> Loud and clear! >>> >>> >>> >>> >>> "Ken Rice" 05/31/17 5:33 PM >>> >>> >>> You got this? >>> >>> >>> >>> From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Brian West >>> Sent: Wednesday, May 31, 2017 3:26 PM >>> To: FreeSWITCH Users Help >>> Subject: Re: [Freeswitch-users] testing 1234 >>> >>> >>> >>> I got this. >>> >>> >>> >>> /b >>> >>> >>> >>> >>> >>> On Wed, May 31, 2017 at 3:22 PM, Ken Rice wrote: >>> >>> This is just a test >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at 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 at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> >>> >>> -- >>> >>> Brian West >>> brian at freeswitch.org >>> >>> Twitter: @FreeSWITCH , @briankwest >>> >>> http://www.freeswitchbook.com >>> http://www.freeswitchcookbook.com >>> >>> Book a phone call (CST) >>> >>> Allison prompts for FreeSWITCH: >>> >>> https://www.gofundme.com/allison-prompts-for-freeswitch >>> >>> Got Bugs? Report them here! | Reddit: /r/freeswitch >>> >>> T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378) >>> Skype:briankwest >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at 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 at 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 at 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 at 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 at 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 at 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 at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > -- > Brian West > brian at freeswitch.org > > Twitter: @FreeSWITCH , @briankwest > > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > Book a phone call (CST) > > Allison prompts for FreeSWITCH: > > https://www.gofundme.com/allison-prompts-for-freeswitch > > Got Bugs? Report them here! | Reddit: /r/freeswitch > > T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378) > Skype:briankwest > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From gb at cm.nl Fri Jun 2 08:54:12 2017 From: gb at cm.nl (Grant Bagdasarian) Date: Fri, 2 Jun 2017 08:54:12 +0000 Subject: [Freeswitch-users] Multiple c params in SDP Message-ID: <7d8d16ed5acb4ada8bc1753fcdae9384@cm.nl> Hi, We're currently facing an issue with one of our clients where they send two c parameters in the SDP. Our Kamailio which is in front of the freeswitch communicates with RTPENGINE which in turn changes only the c parameters in the Media Description, not at Session Level. Freeswitch however, uses the IP in the c param in the Session Description which causes the RTP stream to go directly to the client, instead of being bridged by the RTPENGINE. Is there any way to force freeswitch to use the c param in the Media Description? Regards, Grant -------------- next part -------------- An HTML attachment was scrubbed... URL: From martin at maxnet.ao Fri Jun 2 09:54:57 2017 From: martin at maxnet.ao (Martin Boese) Date: Fri, 2 Jun 2017 10:54:57 +0100 Subject: [Freeswitch-users] Enabling new languages for Say Message-ID: <20170602105457.68aa24f1@bones-tp> Hi! FreeSWITCH Version 1.6.15-32-bec4538~64bit (-32-bec4538 64bit) Debian Jessie. Vanilla config. I am trying to enable portuguese to say numbers using dptools "Say". This is what I did: - Module mod_say_pt is loaded - in freeswitch.xml include lang/pt/pt_PT.xml - Downloaded sounds from https://github.com/jpawlowski/freeswitch-sounds-ttsand placed them into /usr/share/freeswitch/sounds Directory structure is now like: /pt/tts/google/ascii/16000/... CLI> say_string t.wav pt NUMBER pronounced 123 [ERR] switch_xml.c:3180 Can't find phrases tag ..I found out that lang/pt_PT.xml seems to be missing the tags within the tag (vanilla config). I fixed that. Now: CLI> say_string t.wav pt NUMBER pronounced 123 [ERR] switch_ivr.c:3726 Invalid SAY Interface [pt]! BTW: English works fine: CLI> say_string t.wav en NUMBER pronounced 123 file_string://digits/1.wav!digits/hundred.wav!digits/20.wav!digits/3.wav I also tried other languages but have same error "Invalid SAY Interface". What am I missing. Please help. Thanks, Martin From xxxman2008 at 126.com Fri Jun 2 14:26:08 2017 From: xxxman2008 at 126.com (Raymond) Date: Fri, 2 Jun 2017 22:26:08 +0800 (CST) Subject: [Freeswitch-users] question about HA solution In-Reply-To: References: Message-ID: <2628fa1d.b9a5.15c6932ffc0.Coremail.xxxman2008@126.com> Hi, Denys Talking about HA ,it's complex , and have lots of detail. The "FS HA Solution" is very simple . if you think deeper ,you will find more question,such as: A. if we have 5 servers in the group , what is the rule for transfer the calls ? B. Once the broken server have 1000 concurrent call , if there's new "performance issue" when we transfer so many calls to a server. So, forgot your question , the "HA Solution" is just a demo . It need more work to use for business . Raymond 在 2017-06-01 06:28:42，"Denys Pozniak" 写道： Hello! I built FS HA solution based on keepalived and mysql master-master. It works ok generally, but as I understand FS after restarting cleaning own database. So when node1 fails calls jump to node2, after script restarts node1 it is not possible to move calls back. Tried options in switch.conf.xml, but no luck: Is there is a way to solve this? BR, Denys -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian at freeswitch.org Fri Jun 2 14:49:31 2017 From: brian at freeswitch.org (Brian West) Date: Fri, 2 Jun 2017 09:49:31 -0500 Subject: [Freeswitch-users] Multiple c params in SDP In-Reply-To: <7d8d16ed5acb4ada8bc1753fcdae9384@cm.nl> References: <7d8d16ed5acb4ada8bc1753fcdae9384@cm.nl> Message-ID: I seen your post on reddit about this issue. Can you show me traces of what exactly its doing? /b On Fri, Jun 2, 2017 at 3:54 AM, Grant Bagdasarian wrote: > Hi, > > > > We’re currently facing an issue with one of our clients where they send > two c parameters in the SDP. > > Our Kamailio which is in front of the freeswitch communicates with > RTPENGINE which in turn changes only the c parameters in the Media > Description, not at Session Level. > > Freeswitch however, uses the IP in the c param in the Session Description > which causes the RTP stream to go directly to the client, instead of being > bridged by the RTPENGINE. > > > > Is there any way to force freeswitch to use the c param in the Media > Description? > > > Regards, > > > > Grant > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Book a phone call (CST) Allison prompts for FreeSWITCH: *https://www.gofundme.com/allison-prompts-for-freeswitch* Got Bugs? Report them here ! | Reddit: /r/freeswitch *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian at freeswitch.org Fri Jun 2 14:51:25 2017 From: brian at freeswitch.org (Brian West) Date: Fri, 2 Jun 2017 09:51:25 -0500 Subject: [Freeswitch-users] Enabling new languages for Say In-Reply-To: <20170602105457.68aa24f1@bones-tp> References: <20170602105457.68aa24f1@bones-tp> Message-ID: I would guess mod_say_pt is not loaded. On Fri, Jun 2, 2017 at 4:54 AM, Martin Boese wrote: > Hi! > > FreeSWITCH Version 1.6.15-32-bec4538~64bit (-32-bec4538 64bit) > Debian Jessie. Vanilla config. > > I am trying to enable portuguese to say numbers using dptools "Say". > > This is what I did: > - Module mod_say_pt is loaded > - in freeswitch.xml > include lang/pt/pt_PT.xml > - Downloaded sounds from > https://github.com/jpawlowski/freeswitch-sounds-ttsand placed them > into /usr/share/freeswitch/sounds > Directory structure is now like: /pt/tts/google/ascii/16000/... > > CLI> say_string t.wav pt NUMBER pronounced 123 > [ERR] switch_xml.c:3180 Can't find phrases tag > > ..I found out that lang/pt_PT.xml seems to be missing the > tags within the tag (vanilla config). I > fixed that. > > Now: > CLI> say_string t.wav pt NUMBER pronounced 123 > [ERR] switch_ivr.c:3726 Invalid SAY Interface [pt]! > > BTW: English works fine: > CLI> say_string t.wav en NUMBER pronounced 123 > file_string://digits/1.wav!digits/hundred.wav!digits/20.wav!digits/3.wav > > I also tried other languages but have same error "Invalid SAY > Interface". > > What am I missing. Please help. > > Thanks, > Martin > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Book a phone call (CST) Allison prompts for FreeSWITCH: *https://www.gofundme.com/allison-prompts-for-freeswitch* Got Bugs? Report them here ! | Reddit: /r/freeswitch *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: From mjlopez at smartic.es Thu Jun 1 16:00:38 2017 From: mjlopez at smartic.es (=?iso-8859-1?Q?Miguel_Jes=FAs_L=F3pez_Valverde?=) Date: Thu, 1 Jun 2017 18:00:38 +0200 Subject: [Freeswitch-users] trouble with instalation of mod_fail2ban module in FreeSwitch. Message-ID: <010a01d2daf0$36c6cc50$a45464f0$@smartic.es> Hello: I am sending you this email because I am having problems installing the module mod_fail2ban following the recipe offered at https://freeswitch.org/confluence/display/FREESWITCH/mod_fail2ban because when I run make, I get this result: /usr/local/freeswitch/mod/applications/mod_fail2ban$ sudo make Makefile:2: ../../../../build/modmake.rules: No such file or directory make: *** No rule to make target '../../../../build/modmake.rules'. Stop. Do you know in which folder the sentence “git clone” may to be executed?. Do you know if there is anything else to keep in mind that can cause this problem? Thank you very much. Miguel J. Lopez. --- El software de antivirus Avast ha analizado este correo electrónico en busca de virus. https://www.avast.com/antivirus -------------- next part -------------- An HTML attachment was scrubbed... URL: From luis.daniel.lucio at gmail.com Fri Jun 2 15:30:03 2017 From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz) Date: Fri, 2 Jun 2017 11:30:03 -0400 Subject: [Freeswitch-users] Not honoring ODBC in the core Message-ID: my FS 1.6.17 is not honoring my ODBC settings, it keeps creating core.db and other sqlite db's. I can confirm ODBC works, isql command connects without issues. any ideas? -- Luis Daniel Lucio Quiroz CISSP, CISM, CISA Linux, VoIP and much more fun www.okay.com.mx Need LCR? Check out LCR for FusionPBX with FreeSWITCH Need Billing? Check out Billing for FusionPBX with FreeSWITCH -------------- next part -------------- An HTML attachment was scrubbed... URL: From infos at madovsky.org Fri Jun 2 21:58:55 2017 From: infos at madovsky.org (Madovsky) Date: Fri, 2 Jun 2017 14:58:55 -0700 Subject: [Freeswitch-users] trouble with instalation of mod_fail2ban module in FreeSwitch. In-Reply-To: <010a01d2daf0$36c6cc50$a45464f0$@smartic.es> References: <010a01d2daf0$36c6cc50$a45464f0$@smartic.es> Message-ID: <52bacbc4-bc14-e9c9-d938-da1cc789f357@madovsky.org> On 6/1/2017 9:00 AM, Miguel Jesús López Valverde wrote: > > Hello: > > I am sending you this email because I am having problems installing > the module mod_fail2ban following the recipe offered at > https://freeswitch.org/confluence/display/FREESWITCH/mod_fail2ban > > because when I run make, I get this result: > > /usr/local/freeswitch/mod/applications/mod_fail2ban$ sudo make > > Makefile:2: ../../../../build/modmake.rules: No such file or directory > > make: *** No rule to make target '../../../../build/modmake.rules'. Stop. > > Do you know in which folder the sentence “git clone” may to be > executed?. Do you know if there is anything else to keep in mind that > can cause this problem? > > Thank you very much. > > Miguel J. Lopez. > > > > Libre de virus. www.avast.com > > > > <#DAB4FAD8-2DD7-40BB-A1B8-4E2AA1F9FDF2> > > did you install fail2ban from your distro? -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Fri Jun 2 22:07:59 2017 From: mike at jerris.com (Michael Jerris) Date: Fri, 2 Jun 2017 18:07:59 -0400 Subject: [Freeswitch-users] Not honoring ODBC in the core In-Reply-To: References: Message-ID: take a look at the logs to see why it can’t connect to the db. > On Jun 2, 2017, at 11:30 AM, Luis Daniel Lucio Quiroz wrote: > > my FS 1.6.17 is not honoring my ODBC settings, it keeps creating core.db and other sqlite db's. I can confirm ODBC works, isql command connects without issues. > > > > From mike at jerris.com Fri Jun 2 22:09:18 2017 From: mike at jerris.com (Michael Jerris) Date: Fri, 2 Jun 2017 18:09:18 -0400 Subject: [Freeswitch-users] trouble with instalation of mod_fail2ban module in FreeSwitch. In-Reply-To: <010a01d2daf0$36c6cc50$a45464f0$@smartic.es> References: <010a01d2daf0$36c6cc50$a45464f0$@smartic.es> Message-ID: <84E0A820-13D2-47D7-9F7C-5AB5FB0617AC@jerris.com> This is a new module only available in the master branch > On Jun 1, 2017, at 12:00 PM, Miguel Jesús López Valverde wrote: > > Hello: > > I am sending you this email because I am having problems installing the module mod_fail2ban following the recipe offered at https://freeswitch.org/confluence/display/FREESWITCH/mod_fail2ban > because when I run make, I get this result: > > /usr/local/freeswitch/mod/applications/mod_fail2ban$ sudo make > Makefile:2: ../../../../build/modmake.rules: No such file or directory > make: *** No rule to make target '../../../../build/modmake.rules'. Stop. > > Do you know in which folder the sentence “git clone” may to be executed?. Do you know if there is anything else to keep in mind that can cause this problem? > -------------- next part -------------- An HTML attachment was scrubbed... URL: From xxxman2008 at 126.com Sat Jun 3 05:58:08 2017 From: xxxman2008 at 126.com (Raymond) Date: Sat, 3 Jun 2017 13:58:08 +0800 (CST) Subject: [Freeswitch-users] question about HA solution In-Reply-To: References: Message-ID: <4c2d3f0d.1f3e.15c6c8841fa.Coremail.xxxman2008@126.com> Hi, Denys Talking about HA ,it's complex , and have lots of detail. The "FS HA Solution" is very simple . if you think deeper ,you will find more question,such as: A. if we have 5 servers in the group , what is the rule for transfer the calls ? B. Once the broken server have 1000 concurrent call , if there's new "performance issue" when we transfer so many calls to a server. If it really need an answer about your question -- "if it is possible to move calls back". I think it's unnecessary. Every time when we transfer the calls from one server to another, there's a little "audio dropped" , maybe , not good for user experience. So, forgot your question , the "HA Solution" is just a demo . It need more work to do for product situation. Raymond 在 2017-06-01 06:28:42，"Denys Pozniak" 写道： Hello! I built FS HA solution based on keepalived and mysql master-master. It works ok generally, but as I understand FS after restarting cleaning own database. So when node1 fails calls jump to node2, after script restarts node1 it is not possible to move calls back. Tried options in switch.conf.xml, but no luck: Is there is a way to solve this? BR, Denys -------------- next part -------------- An HTML attachment was scrubbed... URL: From donguyenha at gmail.com Sat Jun 3 06:11:37 2017 From: donguyenha at gmail.com (Do Nguyen Ha) Date: Sat, 3 Jun 2017 13:11:37 +0700 Subject: [Freeswitch-users] question about HA solution In-Reply-To: <4c2d3f0d.1f3e.15c6c8841fa.Coremail.xxxman2008@126.com> References: <4c2d3f0d.1f3e.15c6c8841fa.Coremail.xxxman2008@126.com> Message-ID: +1 for Raymond On Jun 3, 2017 12:58, "Raymond" wrote: > Hi, Denys > > Talking about HA ,it's complex , and have lots of detail. > The "FS HA Solution" is very simple . if you think deeper ,you will > find more question,such as: > A. if we have 5 servers in the group , what is the rule for > transfer the calls ? > B. Once the broken server have 1000 concurrent call , if there's > new "performance issue" when we transfer so many calls to a server. > > If it really need an answer about your question -- "if it is possible to > move calls back". I think it's unnecessary. Every time when we transfer > the calls from one server to another, there's a little "audio dropped" , > maybe , not good for user experience. > > So, forgot your question , the "HA Solution" is just a demo . It need > more work to do for product situation. > > Raymond > > > > > 在 2017-06-01 06:28:42，"Denys Pozniak" 写道： > > Hello! > > I built FS HA solution based on keepalived and mysql master-master. > It works ok generally, but as I understand FS after restarting cleaning > own database. > > So when node1 fails calls jump to node2, after script restarts node1 it is > not possible to move calls back. > > Tried options in switch.conf.xml, but no luck: > > > > > Is there is a way to solve this? > > BR, > Denys > > > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From alexdruzhilov at gmail.com Sat Jun 3 07:35:24 2017 From: alexdruzhilov at gmail.com (=?UTF-8?B?0JDQu9C10LrRgdCw0L3QtNGAINCU0YDRg9C20LjQu9C+0LI=?=) Date: Sat, 3 Jun 2017 10:35:24 +0300 Subject: [Freeswitch-users] Fwd: Video conference with transcode mode is not working more than for 5-6 users In-Reply-To: References: Message-ID: OS: CentOS 7.2.1511 Freeswitch: 1.6.17 Steps: 1) create video conference (mod_sofia and conference-mode = transcode) with one video stream from conference owner 2) add 4-6 members in this conference who will receive owner's video stream 3) everyone in this conference will see dramatic degradation of video stream (incoming video stream bitrate falls from 1 Mbps to 0.5 Mbps, frame rate falls down to 10 fps from 30 fps, lag between video and audio stream appears) But I don't see any problems with CPU, memory or network. So does anybody knows whether it is an issue or it's how freeswitch 'transcode' mode should works? And what to do to tune performance of this mode? -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Sat Jun 3 13:57:55 2017 From: mike at jerris.com (Michael Jerris) Date: Sat, 03 Jun 2017 13:57:55 +0000 Subject: [Freeswitch-users] Fwd: Video conference with transcode mode is not working more than for 5-6 users In-Reply-To: References: Message-ID: We develop on debian 8. It's worth trying there to see if it's an issue with versions of things in centos. On Sat, Jun 3, 2017 at 3:37 AM Александр Дружилов wrote: > OS: CentOS 7.2.1511 > Freeswitch: 1.6.17 > > Steps: > 1) create video conference (mod_sofia and conference-mode = transcode) > with one video stream from conference owner > 2) add 4-6 members in this conference who will receive owner's video stream > 3) everyone in this conference will see dramatic degradation of video > stream (incoming video stream bitrate falls from 1 Mbps to 0.5 Mbps, frame > rate falls down to 10 fps from 30 fps, lag between video and audio stream > appears) > > But I don't see any problems with CPU, memory or network. So does anybody > knows whether it is an issue or it's how freeswitch 'transcode' mode should > works? And what to do to tune performance of this mode? > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From anthony.minessale at gmail.com Sat Jun 3 16:26:46 2017 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 03 Jun 2017 16:26:46 +0000 Subject: [Freeswitch-users] Fwd: Video conference with transcode mode is not working more than for 5-6 users In-Reply-To: References: Message-ID: Mux mode with 1x1 layout and minimize-video-encoding flag is better On Sat, Jun 3, 2017 at 8:59 AM Michael Jerris wrote: > We develop on debian 8. It's worth trying there > to see if it's an issue with versions of things in centos. > > On Sat, Jun 3, 2017 at 3:37 AM Александр Дружилов > wrote: > >> OS: CentOS 7.2.1511 >> Freeswitch: 1.6.17 >> >> Steps: >> 1) create video conference (mod_sofia and conference-mode = transcode) >> with one video stream from conference owner >> 2) add 4-6 members in this conference who will receive owner's video >> stream >> 3) everyone in this conference will see dramatic degradation of video >> stream (incoming video stream bitrate falls from 1 Mbps to 0.5 Mbps, frame >> rate falls down to 10 fps from 30 fps, lag between video and audio stream >> appears) >> >> But I don't see any problems with CPU, memory or network. So does anybody >> knows whether it is an issue or it's how freeswitch 'transcode' mode should >> works? And what to do to tune performance of this mode? >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at 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 at 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 at 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 ♬ @anthmfs ♬ @FreeSWITCH ♬ ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ * ClueCon Weekly Development Call ☎ sip:888 at conference.freeswitch.org ☎ +19193869900 https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA -------------- next part -------------- An HTML attachment was scrubbed... URL: From d.mordovin at dwide.com Sun Jun 4 08:10:13 2017 From: d.mordovin at dwide.com (Dmitry Mordovin) Date: Sun, 4 Jun 2017 11:10:13 +0300 Subject: [Freeswitch-users] DTMF events Message-ID: <91ba4d3c-b0a4-a255-2b17-75d91b7767fb@dwide.com> Hello! I want make application which will listen DTMF events and when it fire, send DTMF digit to web-url. For example, I use API for handle call state, execute_on_originate, execute_on_ring, execute_on_answer... Does exists API for DTMF? Anyone knows how can do it? Best regards, Dmitry Mordovin From luis.daniel.lucio at gmail.com Sat Jun 3 20:16:49 2017 From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz) Date: Sat, 3 Jun 2017 16:16:49 -0400 Subject: [Freeswitch-users] question about HA solution In-Reply-To: References: Message-ID: You may want to read this article. http://inside-out.xyz/technology/how-to-configure-freeswitch-for-ha.html Le 31 mai 2017 6:29 PM, "Denys Pozniak" a écrit : Hello! I built FS HA solution based on keepalived and mysql master-master. It works ok generally, but as I understand FS after restarting cleaning own database. So when node1 fails calls jump to node2, after script restarts node1 it is not possible to move calls back. Tried options in switch.conf.xml, but no luck: Is there is a way to solve this? BR, Denys _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From luis.daniel.lucio at gmail.com Sun Jun 4 00:25:34 2017 From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz) Date: Sat, 3 Jun 2017 20:25:34 -0400 Subject: [Freeswitch-users] Not honoring ODBC in the core In-Reply-To: References: Message-ID: I have managed to move all modules to ODBC but the core. If it helps, this deployment is using AWS RDS. I have turned on all the debug options, and i get the following. Please note that the core.db file is still being created. This output is not telling me what is missing or failing to fix it :( Any pointer will be appreciated. 2017-06-04 00:21:52.739199 [INFO] switch_event.c:685 Activate Eventing Engine. 2017-06-04 00:21:52.749458 [WARNING] switch_event.c:656 Create additional event dispatch thread 0 2017-06-04 00:21:52.783529 [INFO] switch_nat.c:417 Scanning for NAT 2017-06-04 00:21:52.783645 [DEBUG] switch_nat.c:170 Checking for PMP 1/5 2017-06-04 00:21:53.783805 [DEBUG] switch_nat.c:170 Checking for PMP 2/5 2017-06-04 00:21:54.783954 [DEBUG] switch_nat.c:170 Checking for PMP 3/5 2017-06-04 00:21:55.784111 [DEBUG] switch_nat.c:170 Checking for PMP 4/5 2017-06-04 00:21:56.784274 [DEBUG] switch_nat.c:170 Checking for PMP 5/5 2017-06-04 00:21:57.784405 [ERR] switch_nat.c:199 Error checking for PMP [general error] 2017-06-04 00:21:57.784435 [DEBUG] switch_nat.c:422 Checking for UPnP 2017-06-04 00:22:09.785033 [INFO] switch_nat.c:438 No PMP or UPnP NAT devices detected! 2017-06-04 00:22:09.786686 [NOTICE] switch_core.c:2326 Set switchname to fs02.prostarsentertainment.com 2017-06-04 00:22:09.787097 [INFO] switch_core_sqldb.c:3396 Opening DB 2017-06-04 00:22:09.787131 [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR [no such table: channels] drop table channels 2017-06-04 00:22:09.787149 [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR [no such table: calls] drop table calls 2017-06-04 00:22:09.787164 [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR [no such view: detailed_calls] drop view detailed_calls 2017-06-04 00:22:09.787179 [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR [no such view: basic_calls] drop view basic_calls 2017-06-04 00:22:09.787193 [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR [no such table: interfaces] drop table interfaces 2017-06-04 00:22:09.787218 [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR [no such table: tasks] drop table tasks 2017-06-04 00:22:09.787393 [DEBUG] switch_core_sqldb.c:1347 SQL ERR [no such table: aliases] [select hostname from aliases] Auto Generating Table! 2017-06-04 00:22:09.787412 [DEBUG] switch_core_sqldb.c:1354 Ignoring SQL ERR [no such table: aliases] [DROP TABLE aliases] 2017-06-04 00:22:09.787562 [DEBUG] switch_core_sqldb.c:1347 SQL ERR [no such table: complete] [select hostname from complete] Auto Generating Table! 2017-06-04 00:22:09.787582 [DEBUG] switch_core_sqldb.c:1354 Ignoring SQL ERR [no such table: complete] [DROP TABLE complete] 2017-06-04 00:22:09.787709 [DEBUG] switch_core_sqldb.c:1347 SQL ERR [no such table: nat] [select hostname from nat] Auto Generating Table! 2017-06-04 00:22:09.787728 [DEBUG] switch_core_sqldb.c:1354 Ignoring SQL ERR [no such table: nat] [DROP TABLE nat] 2017-06-04 00:22:09.787822 [DEBUG] switch_core_sqldb.c:1347 SQL ERR [no such table: registrations] [delete from registrations where reg_user=''] Auto Generating Table! 2017-06-04 00:22:09.787842 [DEBUG] switch_core_sqldb.c:1354 Ignoring SQL ERR [no such table: registrations] [DROP TABLE registrations] 2017-06-04 00:22:09.788012 [DEBUG] switch_core_sqldb.c:1347 SQL ERR [no such table: recovery] [select hostname from recovery] Auto Generating Table! 2017-06-04 00:22:09.788033 [DEBUG] switch_core_sqldb.c:1354 Ignoring SQL ERR [no such table: recovery] [DROP TABLE recovery] 2017-06-04 00:22:09.792959 [INFO] switch_core_sqldb.c:1693 CORE Starting SQL thread. 2017-06-04 00:22:09.797506 [DEBUG] switch_scheduler.c:249 Added task 1 heartbeat (core) to run at 1496535729 2017-06-04 00:22:09.797533 [DEBUG] switch_scheduler.c:249 Added task 2 check_ip (core) to run at 1496535729 -- Luis Daniel Lucio Quiroz CISSP, CISM, CISA Linux, VoIP and much more fun www.okay.com.mx Need LCR? Check out LCR for FusionPBX with FreeSWITCH Need Billing? Check out Billing for FusionPBX with FreeSWITCH On Fri, Jun 2, 2017 at 6:07 PM, Michael Jerris wrote: > take a look at the logs to see why it can’t connect to the db. > > > On Jun 2, 2017, at 11:30 AM, Luis Daniel Lucio Quiroz < > luis.daniel.lucio at gmail.com> wrote: > > > > my FS 1.6.17 is not honoring my ODBC settings, it keeps creating core.db > and other sqlite db's. I can confirm ODBC works, isql command connects > without issues. > > > > > > /> > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From luis.daniel.lucio at gmail.com Sun Jun 4 00:57:00 2017 From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz) Date: Sat, 3 Jun 2017 20:57:00 -0400 Subject: [Freeswitch-users] Not honoring ODBC in the core In-Reply-To: References: Message-ID: It is working. It sounds like a dummy error, but after reading the source, specifically the function _switch_cache_db_get_db_handle_dsn in the file src/switch_core_sqldb.c, first i found that it is better to prefix with ,odbc://, second I found in the freeswitch.xml the big issue with , it seems that my syncthing was creating double files without taking out the .xml extension, this makes the to process another "backup" file which it was overwriting the variable. Thanks for your time -- Luis Daniel Lucio Quiroz CISSP, CISM, CISA Linux, VoIP and much more fun www.okay.com.mx Need LCR? Check out LCR for FusionPBX with FreeSWITCH Need Billing? Check out Billing for FusionPBX with FreeSWITCH On Sat, Jun 3, 2017 at 8:25 PM, Luis Daniel Lucio Quiroz < luis.daniel.lucio at gmail.com> wrote: > I have managed to move all modules to ODBC but the core. If it helps, this > deployment is using AWS RDS. > > I have turned on all the debug options, and i get the following. Please > note that the core.db file is still being created. This output is not > telling me what is missing or failing to fix it :( Any pointer will be > appreciated. > > 2017-06-04 00:21:52.739199 [INFO] switch_event.c:685 Activate Eventing > Engine. > 2017-06-04 00:21:52.749458 [WARNING] switch_event.c:656 Create additional > event dispatch thread 0 > 2017-06-04 00:21:52.783529 [INFO] switch_nat.c:417 Scanning for NAT > 2017-06-04 00:21:52.783645 [DEBUG] switch_nat.c:170 Checking for PMP 1/5 > 2017-06-04 00:21:53.783805 [DEBUG] switch_nat.c:170 Checking for PMP 2/5 > 2017-06-04 00:21:54.783954 [DEBUG] switch_nat.c:170 Checking for PMP 3/5 > 2017-06-04 00:21:55.784111 [DEBUG] switch_nat.c:170 Checking for PMP 4/5 > 2017-06-04 00:21:56.784274 [DEBUG] switch_nat.c:170 Checking for PMP 5/5 > 2017-06-04 00:21:57.784405 [ERR] switch_nat.c:199 Error checking for PMP > [general error] > 2017-06-04 00:21:57.784435 [DEBUG] switch_nat.c:422 Checking for UPnP > 2017-06-04 00:22:09.785033 [INFO] switch_nat.c:438 No PMP or UPnP NAT > devices detected! > 2017-06-04 00:22:09.786686 [NOTICE] switch_core.c:2326 Set switchname to > fs02.prostarsentertainment.com > 2017-06-04 00:22:09.787097 [INFO] switch_core_sqldb.c:3396 Opening DB > 2017-06-04 00:22:09.787131 [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR > [no such table: channels] > drop table channels > 2017-06-04 00:22:09.787149 [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR > [no such table: calls] > drop table calls > 2017-06-04 00:22:09.787164 [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR > [no such view: detailed_calls] > drop view detailed_calls > 2017-06-04 00:22:09.787179 [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR > [no such view: basic_calls] > drop view basic_calls > 2017-06-04 00:22:09.787193 [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR > [no such table: interfaces] > drop table interfaces > 2017-06-04 00:22:09.787218 [ERR] switch_core_sqldb.c:587 NATIVE SQL ERR > [no such table: tasks] > drop table tasks > 2017-06-04 00:22:09.787393 [DEBUG] switch_core_sqldb.c:1347 SQL ERR [no > such table: aliases] > [select hostname from aliases] > Auto Generating Table! > 2017-06-04 00:22:09.787412 [DEBUG] switch_core_sqldb.c:1354 Ignoring SQL > ERR [no such table: aliases] > [DROP TABLE aliases] > 2017-06-04 00:22:09.787562 [DEBUG] switch_core_sqldb.c:1347 SQL ERR [no > such table: complete] > [select hostname from complete] > Auto Generating Table! > 2017-06-04 00:22:09.787582 [DEBUG] switch_core_sqldb.c:1354 Ignoring SQL > ERR [no such table: complete] > [DROP TABLE complete] > 2017-06-04 00:22:09.787709 [DEBUG] switch_core_sqldb.c:1347 SQL ERR [no > such table: nat] > [select hostname from nat] > Auto Generating Table! > 2017-06-04 00:22:09.787728 [DEBUG] switch_core_sqldb.c:1354 Ignoring SQL > ERR [no such table: nat] > [DROP TABLE nat] > 2017-06-04 00:22:09.787822 [DEBUG] switch_core_sqldb.c:1347 SQL ERR [no > such table: registrations] > [delete from registrations where reg_user=''] > Auto Generating Table! > 2017-06-04 00:22:09.787842 [DEBUG] switch_core_sqldb.c:1354 Ignoring SQL > ERR [no such table: registrations] > [DROP TABLE registrations] > 2017-06-04 00:22:09.788012 [DEBUG] switch_core_sqldb.c:1347 SQL ERR [no > such table: recovery] > [select hostname from recovery] > Auto Generating Table! > 2017-06-04 00:22:09.788033 [DEBUG] switch_core_sqldb.c:1354 Ignoring SQL > ERR [no such table: recovery] > [DROP TABLE recovery] > 2017-06-04 00:22:09.792959 [INFO] switch_core_sqldb.c:1693 CORE Starting > SQL thread. > 2017-06-04 00:22:09.797506 [DEBUG] switch_scheduler.c:249 Added task 1 > heartbeat (core) to run at 1496535729 > 2017-06-04 00:22:09.797533 [DEBUG] switch_scheduler.c:249 Added task 2 > check_ip (core) to run at 1496535729 > > > > > -- > Luis Daniel Lucio Quiroz > CISSP, CISM, CISA > Linux, VoIP and much more fun > www.okay.com.mx > > Need LCR? Check out LCR for FusionPBX with FreeSWITCH > Need Billing? Check out Billing for FusionPBX with FreeSWITCH > > On Fri, Jun 2, 2017 at 6:07 PM, Michael Jerris wrote: > >> take a look at the logs to see why it can’t connect to the db. >> >> > On Jun 2, 2017, at 11:30 AM, Luis Daniel Lucio Quiroz < >> luis.daniel.lucio at gmail.com> wrote: >> > >> > my FS 1.6.17 is not honoring my ODBC settings, it keeps creating >> core.db and other sqlite db's. I can confirm ODBC works, isql command >> connects without issues. >> > >> > >> > > /> >> > >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From royj at yandex.ru Sun Jun 4 10:22:44 2017 From: royj at yandex.ru (royj at yandex.ru) Date: Sun, 04 Jun 2017 13:22:44 +0300 Subject: [Freeswitch-users] DTMF events In-Reply-To: <91ba4d3c-b0a4-a255-2b17-75d91b7767fb@dwide.com> References: <91ba4d3c-b0a4-a255-2b17-75d91b7767fb@dwide.com> Message-ID: <5532821496571764@web40j.yandex.ru> May be using mod_event_socket ( https://freeswitch.org/confluence/display/FREESWITCH/mod_event_socket ) will be good for you. You can listen for DTMF events arrived and other events withal. 04.06.2017, 11:19, "Dmitry Mordovin" : > Hello! > > I want make application which will listen DTMF events and when it fire, > send DTMF digit to web-url. > > For example, I use API for handle call state, execute_on_originate, > execute_on_ring, execute_on_answer... > > Does exists API for DTMF? > > Anyone knows how can do it? > > Best regards, Dmitry Mordovin > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From jurijs.ivolga at gmail.com Sun Jun 4 11:39:40 2017 From: jurijs.ivolga at gmail.com (Jurijs Ivolga) Date: Sun, 4 Jun 2017 14:39:40 +0300 Subject: [Freeswitch-users] DTMF events In-Reply-To: <5532821496571764@web40j.yandex.ru> References: <91ba4d3c-b0a4-a255-2b17-75d91b7767fb@dwide.com> <5532821496571764@web40j.yandex.ru> Message-ID: Hi, I think this is what you are looking for: https://wiki.freeswitch.org/wiki/Channel_Variables#api_on_tone_detect With kind regards, Jurijs On Sun, Jun 4, 2017 at 1:22 PM, wrote: > May be using mod_event_socket ( https://freeswitch.org/ > confluence/display/FREESWITCH/mod_event_socket ) will be good for you. > You can listen for DTMF events arrived and other events withal. > > 04.06.2017, 11:19, "Dmitry Mordovin" : > > Hello! > > > > I want make application which will listen DTMF events and when it fire, > > send DTMF digit to web-url. > > > > For example, I use API for handle call state, execute_on_originate, > > execute_on_ring, execute_on_answer... > > > > Does exists API for DTMF? > > > > Anyone knows how can do it? > > > > Best regards, Dmitry Mordovin > > > > ____________________________________________________________ > _____________ > > Professional FreeSWITCH Consulting Services: > > consulting at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From alexdruzhilov at gmail.com Mon Jun 5 09:40:08 2017 From: alexdruzhilov at gmail.com (=?UTF-8?B?0JDQu9C10LrRgdCw0L3QtNGAINCU0YDRg9C20LjQu9C+0LI=?=) Date: Mon, 5 Jun 2017 12:40:08 +0300 Subject: [Freeswitch-users] Fwd: Video conference with transcode mode is not working more than for 5-6 users In-Reply-To: References: Message-ID: My use case is to make video conference where anybody can share his video from camera or share screen. Video from camera and screen could have different resolutions and frame rates so I would prefer transcode because mux forces me to use a fixed frame rate and canvas size. 2017-06-03 19:26 GMT+03:00 Anthony Minessale : > Mux mode with 1x1 layout and minimize-video-encoding flag is better > > > On Sat, Jun 3, 2017 at 8:59 AM Michael Jerris wrote: > >> We develop on debian 8. It's worth trying there >> to see if it's an issue with versions of things in centos. >> >> On Sat, Jun 3, 2017 at 3:37 AM Александр Дружилов < >> alexdruzhilov at gmail.com> wrote: >> >>> OS: CentOS 7.2.1511 >>> Freeswitch: 1.6.17 >>> >>> Steps: >>> 1) create video conference (mod_sofia and conference-mode = transcode) >>> with one video stream from conference owner >>> 2) add 4-6 members in this conference who will receive owner's video >>> stream >>> 3) everyone in this conference will see dramatic degradation of video >>> stream (incoming video stream bitrate falls from 1 Mbps to 0.5 Mbps, frame >>> rate falls down to 10 fps from 30 fps, lag between video and audio stream >>> appears) >>> >>> But I don't see any problems with CPU, memory or network. So does >>> anybody knows whether it is an issue or it's how freeswitch 'transcode' >>> mode should works? And what to do to tune performance of this mode? >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at 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 at 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 at 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 at 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 ♬ @anthmfs ♬ @FreeSWITCH ♬ > > ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ > http://twitter.com/FreeSWITCH > ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ > * > > ClueCon Weekly Development Call > ☎ sip:888 at conference.freeswitch.org ☎ +19193869900 > > https://www.youtube.com/watch?v=9XXgW34t40s > https://www.youtube.com/watch?v=NLaDpGQuZDA > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From loi.dangthanh at gmail.com Mon Jun 5 09:46:46 2017 From: loi.dangthanh at gmail.com (=?UTF-8?B?TOG7o2kgxJDhurduZw==?=) Date: Mon, 05 Jun 2017 09:46:46 +0000 Subject: [Freeswitch-users] About proxy-hold codec re-negotiation behavior. In-Reply-To: References: Message-ID: Hi guys, I resend this because I guess my first email does not reach at all. Helps are appreciated. rgds, Loi Dang On Mon, May 22, 2017 at 5:49 PM Lợi Đặng wrote: > Hi list, I'm using FS with *proxy-hold*. > Call flow is simple > A -> FreeSWITCH -> B > Assume FS, A and B support PCMA and PCMU, FS prefer PCMA in codec-prefs > profile configuration. > In affects of my configuration for *greedy early negotiation* and *disable > transcoding*, PCMA is negotiated in both legs for initial INVITE, that's > good, and expected. > > But then I remove PCMA from A and compose a re-INVITE for holding with > PCMU only, the a leg is re-negotiated with PCMU, but the b leg have > re-INVITE with PCMA due to *proxy-hold* variable, that causes transcoding > happened after hold. > > My desire is to have FS to re-negotiate with b leg too, not so similar but > as what I observed when I was using it with media proxy option ( A-A before > hold, U-U after hold). > > I tried late negotiation in this case, but no luck. > So is that not able for FS to re-negotiate codec on b leg, in using > proxy-hold? > Any advise is appreciated. > > rgds, > Loi Dang > -------------- next part -------------- An HTML attachment was scrubbed... URL: From boesemar at gmail.com Mon Jun 5 09:53:00 2017 From: boesemar at gmail.com (boesemar .) Date: Mon, 5 Jun 2017 10:53:00 +0100 Subject: [Freeswitch-users] Enabling new languages for Say Message-ID: Hi Brian, mod_say_pt is definitely loaded: > module_exists mod_say_pt true Can you think of any other cause for the "Invalid SAY Interface [pt]" ? Martin PS: sorry for breaking the thread - problem with email Brian West brian at freeswitch.org Fri Jun 2 14:51:25 UTC 2017 I would guess mod_say_pt is not loaded. On Fri, Jun 2, 2017 at 4:54 AM, Martin Boese > wrote: >* Hi! *>>* FreeSWITCH Version 1.6.15-32-bec4538~64bit (-32-bec4538 64bit) *>* Debian Jessie. Vanilla config. *>>* I am trying to enable portuguese to say numbers using dptools "Say". *>>* This is what I did: *>* - Module mod_say_pt is loaded *>* - in freeswitch.xml *>* include lang/pt/pt_PT.xml *>* - Downloaded sounds from *>* https://github.com/jpawlowski/freeswitch-sounds-ttsand placed them *>* into /usr/share/freeswitch/sounds *>* Directory structure is now like: /pt/tts/google/ascii/16000/... *>>* CLI> say_string t.wav pt NUMBER pronounced 123 *>* [ERR] switch_xml.c:3180 Can't find phrases tag *>>* ..I found out that lang/pt_PT.xml seems to be missing the *>* tags within the tag (vanilla config). I *>* fixed that. *>>* Now: *>* CLI> say_string t.wav pt NUMBER pronounced 123 *>* [ERR] switch_ivr.c:3726 Invalid SAY Interface [pt]! *>>* BTW: English works fine: *>* CLI> say_string t.wav en NUMBER pronounced 123 *>* file_string://digits/1.wav!digits/hundred.wav!digits/20.wav!digits/3.wav *>>* I also tried other languages but have same error "Invalid SAY *>* Interface". *>>* What am I missing. Please help. *>>* Thanks, *>* Martin *>>* _________________________________________________________________________ *>* Professional FreeSWITCH Consulting Services: *>* consulting at 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 at lists.freeswitch.org *>* http://lists.freeswitch.org/mailman/listinfo/freeswitch-users *>* UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users *>* http://www.freeswitch.org * -- *Brian West*brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.comhttp://www.freeswitchcookbook.com Book a phone call (CST) Allison prompts for FreeSWITCH: *https://www.gofundme.com/allison-prompts-for-freeswitch* Got Bugs? Report them here ! | Reddit: /r/freeswitch *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: ------------------------------ -------------- next part -------------- An HTML attachment was scrubbed... URL: From denys.pozniak at crazycall.com Mon Jun 5 11:41:16 2017 From: denys.pozniak at crazycall.com (Denys Pozniak) Date: Mon, 5 Jun 2017 13:41:16 +0200 Subject: [Freeswitch-users] question about HA solution In-Reply-To: References: Message-ID: Hello! Thank you *Raymond* about your explanation, but I dont agree with some point: *If it really need an answer about your question -- "if it is possible to move calls back". I think it's unnecessary.* - in my case I have two not equal servers, so I need to have only one as a master. If switchover happens I need to have ability to restore master back. Thank you *Luis* for your link, you can do simple test to understand what I am talking about: do call -> check on master and slave #show channels -> restart FS on slave -> check on master #show channels. In my case I dont see any active calls after this, so restoring back is not possible. On 3 June 2017 at 22:16, Luis Daniel Lucio Quiroz < luis.daniel.lucio at gmail.com> wrote: > You may want to read this article. > > http://inside-out.xyz/technology/how-to-configure-freeswitch-for-ha.html > > Le 31 mai 2017 6:29 PM, "Denys Pozniak" a > écrit : > > Hello! > > I built FS HA solution based on keepalived and mysql master-master. > It works ok generally, but as I understand FS after restarting cleaning > own database. > > So when node1 fails calls jump to node2, after script restarts node1 it is > not possible to move calls back. > > Tried options in switch.conf.xml, but no luck: > > > > > Is there is a way to solve this? > > BR, > Denys > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From vladislaus at gmail.com Mon Jun 5 13:20:14 2017 From: vladislaus at gmail.com (Andres Gomez) Date: Mon, 5 Jun 2017 08:20:14 -0500 Subject: [Freeswitch-users] Videoconference error with H263+ [ERR] avcodec.c:736 Message-ID: Hello. I see this console error videcoconference with H263+ 2017-06-05 08:14:44.668608 [ERR] avcodec.c:736 len: 1200, mark:0 00 00 d2 ef 2017-06-05 08:14:44.668608 [ERR] avcodec.c:736 len: 1200, mark:0 00 00 f4 12 2017-06-05 08:14:44.668608 [ERR] avcodec.c:736 len: 382, mark:1 00 00 01 c0 2017-06-05 08:14:44.678607 [ERR] avcodec.c:736 len: 1198, mark:0 04 00 82 ca 2017-06-05 08:14:44.678607 [ERR] avcodec.c:736 len: 1200, mark:0 00 00 56 83 2017-06-05 08:14:44.678607 [ERR] avcodec.c:736 len: 1200, mark:0 00 00 24 4e 2017-06-05 08:14:44.678607 [ERR] avcodec.c:736 len: 1200, mark:0 00 00 e9 db 2017-06-05 08:14:44.678607 [ERR] avcodec.c:736 len: 1200, mark:0 00 00 9a 3a Any Idea why? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian at freeswitch.org Mon Jun 5 14:27:00 2017 From: brian at freeswitch.org (Brian West) Date: Mon, 5 Jun 2017 09:27:00 -0500 Subject: [Freeswitch-users] About proxy-hold codec re-negotiation behavior. In-Reply-To: References: Message-ID: Its working as designed. You have to remember we're a B2BUA... so we have some behaviors that may behave proxy like, but FreeSWITCH is still not a proxy. On Mon, Jun 5, 2017 at 4:46 AM, Lợi Đặng wrote: > Hi guys, I resend this because I guess my first email does not reach at > all. > Helps are appreciated. > rgds, > Loi Dang > > On Mon, May 22, 2017 at 5:49 PM Lợi Đặng wrote: > >> Hi list, I'm using FS with *proxy-hold*. >> Call flow is simple >> A -> FreeSWITCH -> B >> Assume FS, A and B support PCMA and PCMU, FS prefer PCMA in codec-prefs >> profile configuration. >> In affects of my configuration for *greedy early negotiation* and *disable >> transcoding*, PCMA is negotiated in both legs for initial INVITE, that's >> good, and expected. >> >> But then I remove PCMA from A and compose a re-INVITE for holding with >> PCMU only, the a leg is re-negotiated with PCMU, but the b leg have >> re-INVITE with PCMA due to *proxy-hold* variable, that causes >> transcoding happened after hold. >> >> My desire is to have FS to re-negotiate with b leg too, not so similar >> but as what I observed when I was using it with media proxy option ( A-A >> before hold, U-U after hold). >> >> I tried late negotiation in this case, but no luck. >> So is that not able for FS to re-negotiate codec on b leg, in using >> proxy-hold? >> Any advise is appreciated. >> >> rgds, >> Loi Dang >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Book a phone call (CST) Allison prompts for FreeSWITCH: *https://www.gofundme.com/allison-prompts-for-freeswitch* Got Bugs? Report them here ! | Reddit: /r/freeswitch *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Mon Jun 5 15:03:26 2017 From: mike at jerris.com (Michael Jerris) Date: Mon, 5 Jun 2017 11:03:26 -0400 Subject: [Freeswitch-users] Enabling new languages for Say In-Reply-To: References: Message-ID: <590ED298-8358-44F1-B391-B9FCA07AE6DA@jerris.com> what log output do you get when you load mod_say? what output do you get to “show say” > On Jun 5, 2017, at 5:53 AM, boesemar . wrote: > > Hi Brian, > > mod_say_pt is definitely loaded: > > > module_exists mod_say_pt > true > > Can you think of any other cause for the "Invalid SAY Interface [pt]" ? > Martin > > PS: sorry for breaking the thread - problem with email > > > Brian West brian at freeswitch.org > Fri Jun 2 14:51:25 UTC 2017 > > I would guess mod_say_pt is not loaded. > > On Fri, Jun 2, 2017 at 4:54 AM, Martin Boese > wrote: > > > Hi! > > > > FreeSWITCH Version 1.6.15-32-bec4538~64bit (-32-bec4538 64bit) > > Debian Jessie. Vanilla config. > > > > I am trying to enable portuguese to say numbers using dptools "Say". > > > > This is what I did: > > - Module mod_say_pt is loaded > > - in freeswitch.xml > > include lang/pt/pt_PT.xml > > - Downloaded sounds from > > https://github.com/jpawlowski/freeswitch-sounds-ttsand placed them > > into /usr/share/freeswitch/sounds > > Directory structure is now like: /pt/tts/google/ascii/16000/... > > > > CLI> say_string t.wav pt NUMBER pronounced 123 > > [ERR] switch_xml.c:3180 Can't find phrases tag > > > > ..I found out that lang/pt_PT.xml seems to be missing the > > tags within the tag (vanilla config). I > > fixed that. > > > > Now: > > CLI> say_string t.wav pt NUMBER pronounced 123 > > [ERR] switch_ivr.c:3726 Invalid SAY Interface [pt]! > > > > BTW: English works fine: > > CLI> say_string t.wav en NUMBER pronounced 123 > > file_string://digits/1.wav!digits/hundred.wav!digits/20.wav!digits/3.wav <> > > > > I also tried other languages but have same error "Invalid SAY > > Interface". > > > > What am I missing. Please help. > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Mon Jun 5 15:04:25 2017 From: mike at jerris.com (Michael Jerris) Date: Mon, 5 Jun 2017 11:04:25 -0400 Subject: [Freeswitch-users] question about HA solution In-Reply-To: References: Message-ID: <8A13E0EC-FA13-4267-80F6-CE1A8E8360CF@jerris.com> recovered calls will get new entries in the table. > On Jun 5, 2017, at 7:41 AM, Denys Pozniak wrote: > > Hello! > > Thank you Raymond about your explanation, but I dont agree with some point: > If it really need an answer about your question -- "if it is possible to move calls back". I think it's unnecessary. - in my case I have two not equal servers, so I need to have only one as a master. > If switchover happens I need to have ability to restore master back. > > Thank you Luis for your link, you can do simple test to understand what I am talking about: do call -> check on master and slave #show channels -> restart FS on slave -> check on master #show channels. In my case I dont see any active calls after this, so restoring back is not possible. > > > > On 3 June 2017 at 22:16, Luis Daniel Lucio Quiroz > wrote: > You may want to read this article. > > http://inside-out.xyz/technology/how-to-configure-freeswitch-for-ha.html > > Le 31 mai 2017 6:29 PM, "Denys Pozniak" > a écrit : > Hello! > > I built FS HA solution based on keepalived and mysql master-master. > It works ok generally, but as I understand FS after restarting cleaning own database. > > So when node1 fails calls jump to node2, after script restarts node1 it is not possible to move calls back. > > Tried options in switch.conf.xml, but no luck: > > > > > Is there is a way to solve this? > -------------- next part -------------- An HTML attachment was scrubbed... URL: From denys.pozniak at crazycall.com Mon Jun 5 17:50:18 2017 From: denys.pozniak at crazycall.com (Denys Pozniak) Date: Mon, 5 Jun 2017 19:50:18 +0200 Subject: [Freeswitch-users] question about HA solution In-Reply-To: <8A13E0EC-FA13-4267-80F6-CE1A8E8360CF@jerris.com> References: <8A13E0EC-FA13-4267-80F6-CE1A8E8360CF@jerris.com> Message-ID: Yes, correct. But when you restart FS on slave, it will erase database. And option auto-clear-sql=false not working for me. On Jun 5, 2017 6:32 PM, "Michael Jerris" wrote: recovered calls will get new entries in the table. On Jun 5, 2017, at 7:41 AM, Denys Pozniak wrote: Hello! Thank you *Raymond* about your explanation, but I dont agree with some point: *If it really need an answer about your question -- "if it is possible to move calls back". I think it's unnecessary.* - in my case I have two not equal servers, so I need to have only one as a master. If switchover happens I need to have ability to restore master back. Thank you *Luis* for your link, you can do simple test to understand what I am talking about: do call -> check on master and slave #show channels -> restart FS on slave -> check on master #show channels. In my case I dont see any active calls after this, so restoring back is not possible. On 3 June 2017 at 22:16, Luis Daniel Lucio Quiroz < luis.daniel.lucio at gmail.com> wrote: > You may want to read this article. > > http://inside-out.xyz/technology/how-to-configure-freeswitch-for-ha.html > > Le 31 mai 2017 6:29 PM, "Denys Pozniak" a > écrit : > > Hello! > > I built FS HA solution based on keepalived and mysql master-master. > It works ok generally, but as I understand FS after restarting cleaning > own database. > > So when node1 fails calls jump to node2, after script restarts node1 it is > not possible to move calls back. > > Tried options in switch.conf.xml, but no luck: > > > > > Is there is a way to solve this? > > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian at freeswitch.org Mon Jun 5 19:39:03 2017 From: brian at freeswitch.org (Brian West) Date: Mon, 5 Jun 2017 14:39:03 -0500 Subject: [Freeswitch-users] Calling on the community for Bug Marshals Message-ID: FreeSWITCHers, We are in need of a few good bug marshals, We are trying to get 1.8 ready and out the door and the more help we have testing and working thru patches on JIRA the quicker it will arrive. If you're interested in helping us out email me directly. We are also considering bringing back a few days a week we are sitting in 888 and helping the community out with issues pending in JIRA. Also we are only about 2600 short on the gofund me for the Allison prompts, which will be delivered sometime this week. ;) So help us get over that last little bit this week. Thanks, -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Book a phone call (CST) Allison prompts for FreeSWITCH: *https://www.gofundme.com/allison-prompts-for-freeswitch* Got Bugs? Report them here ! | Reddit: /r/freeswitch *T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002 <(918)%20420-9002> | *M:*+1918424WEST (9378) *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: From loi.dangthanh at gmail.com Tue Jun 6 02:31:28 2017 From: loi.dangthanh at gmail.com (=?UTF-8?B?TOG7o2kgxJDhurduZw==?=) Date: Tue, 06 Jun 2017 02:31:28 +0000 Subject: [Freeswitch-users] About proxy-hold codec re-negotiation behavior. In-Reply-To: References: Message-ID: yep, I'm not expecting too much this can be done, the design is good but I'm just curious about profile variables `renegotiate-codec-on-reinvite/hold` that was removed since 1.6.10, guess the job is done in old days. rgds, Loi Dang On Mon, Jun 5, 2017 at 9:30 PM Brian West wrote: > Its working as designed. You have to remember we're a B2BUA... so we have > some behaviors that may behave proxy like, but FreeSWITCH is still not a > proxy. > > On Mon, Jun 5, 2017 at 4:46 AM, Lợi Đặng wrote: > >> Hi guys, I resend this because I guess my first email does not reach at >> all. >> Helps are appreciated. >> rgds, >> Loi Dang >> >> On Mon, May 22, 2017 at 5:49 PM Lợi Đặng wrote: >> >>> Hi list, I'm using FS with *proxy-hold*. >>> Call flow is simple >>> A -> FreeSWITCH -> B >>> Assume FS, A and B support PCMA and PCMU, FS prefer PCMA in codec-prefs >>> profile configuration. >>> In affects of my configuration for *greedy early negotiation* and *disable >>> transcoding*, PCMA is negotiated in both legs for initial INVITE, >>> that's good, and expected. >>> >>> But then I remove PCMA from A and compose a re-INVITE for holding with >>> PCMU only, the a leg is re-negotiated with PCMU, but the b leg have >>> re-INVITE with PCMA due to *proxy-hold* variable, that causes >>> transcoding happened after hold. >>> >>> My desire is to have FS to re-negotiate with b leg too, not so similar >>> but as what I observed when I was using it with media proxy option ( A-A >>> before hold, U-U after hold). >>> >>> I tried late negotiation in this case, but no luck. >>> So is that not able for FS to re-negotiate codec on b leg, in using >>> proxy-hold? >>> Any advise is appreciated. >>> >>> rgds, >>> Loi Dang >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > > *Brian West* > brian at freeswitch.org > > *Twitter: @FreeSWITCH , @briankwest* > > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > Book a phone call (CST) > > Allison prompts for FreeSWITCH: > > *https://www.gofundme.com/allison-prompts-for-freeswitch* > > > Got Bugs? Report them here ! | Reddit: > /r/freeswitch > > *T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002 <(918)%20420-9002> > | *M:*+1918424WEST (9378) > *Skype:*briankwest > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From hunterj91 at hotmail.com Tue Jun 6 06:59:43 2017 From: hunterj91 at hotmail.com (Jonathan Hunter) Date: Tue, 6 Jun 2017 06:59:43 +0000 Subject: [Freeswitch-users] Monitoring call quality using variable_rtp_audio_xxx In-Reply-To: References: Message-ID: Hi Guys, To be more specific, what is the meaning of this parameter? in_flaw_total I am becoming clear on the other parameters! Thanks Jon ________________________________ From: Jonathan Hunter Sent: 31 May 2017 19:24 To: freeswitch-users at lists.freeswitch.org Subject: Monitoring call quality using variable_rtp_audio_xxx Hi Guys, Sorry for the noise, we are looking to poll calls in progress to grab the rtp_audio variables when a call is in progress, and we want to understand if this is a good approach and to that end what values/variables we should consider and what ranges should we be working with? I understand mos and quality percentage but what other values are a good indicator? Again Im clear on jitter and its meaning just want to understand whats what as not everything understandably is not documented. Many thanks Jon "variable_rtp_audio_recv_pt": "8", "variable_rtp_audio_in_raw_bytes": "374960", "variable_rtp_audio_in_media_bytes": "374616", "variable_rtp_audio_in_packet_count": "2180", "variable_rtp_audio_in_media_packet_count": "2178", "variable_rtp_audio_in_skip_packet_count": "92", "variable_rtp_audio_in_jitter_packet_count": "0", "variable_rtp_audio_in_dtmf_packet_count": "0", "variable_rtp_audio_in_cng_packet_count": "0", "variable_rtp_audio_in_flush_packet_count": "2", "variable_rtp_audio_in_largest_jb_size": "0", "variable_rtp_audio_in_jitter_min_variance": "28.57", "variable_rtp_audio_in_jitter_max_variance": "116.33", "variable_rtp_audio_in_jitter_loss_rate": "0.02", "variable_rtp_audio_in_jitter_burst_rate": "0.98", "variable_rtp_audio_in_mean_interval": "20.39", "variable_rtp_audio_in_flaw_total": "43", "variable_rtp_audio_in_quality_percentage": "97.00", "variable_rtp_audio_in_mos": "4.47", "variable_rtp_audio_out_raw_bytes": "210012", "variable_rtp_audio_out_media_bytes": "210012", "variable_rtp_audio_out_packet_count": "1221", "variable_rtp_audio_out_media_packet_count": "1221", "variable_rtp_audio_out_skip_packet_count": "0", "variable_rtp_audio_out_dtmf_packet_count": "0", "variable_rtp_audio_out_cng_packet_count": "0", "variable_rtp_audio_rtcp_packet_count": "0", "variable_rtp_audio_rtcp_octet_count": "0", -------------- next part -------------- An HTML attachment was scrubbed... URL: From boesemar at gmail.com Tue Jun 6 07:19:38 2017 From: boesemar at gmail.com (boesemar .) Date: Tue, 6 Jun 2017 08:19:38 +0100 Subject: [Freeswitch-users] Enabling new languages for Say In-Reply-To: <590ED298-8358-44F1-B391-B9FCA07AE6DA@jerris.com> References: <590ED298-8358-44F1-B391-B9FCA07AE6DA@jerris.com> Message-ID: Hi Michael, > show say type,name,ikey say,en,mod_say_en say,pt,mod_say_pt 2 total. Thx, Martin On Mon, Jun 5, 2017 at 4:03 PM, Michael Jerris wrote: > what log output do you get when you load mod_say? what output do you get > to “show say” > > On Jun 5, 2017, at 5:53 AM, boesemar . wrote: > > Hi Brian, > > mod_say_pt is definitely loaded: > > > module_exists mod_say_pt > true > > Can you think of any other cause for the "Invalid SAY Interface [pt]" ? > > Martin > > > PS: sorry for breaking the thread - problem with email > > > > Brian West brian at freeswitch.org > Fri Jun 2 14:51:25 UTC 2017 > > I would guess mod_say_pt is not loaded. > > On Fri, Jun 2, 2017 at 4:54 AM, Martin Boese > wrote: > > >* Hi! > *>>* FreeSWITCH Version 1.6.15-32-bec4538~64bit (-32-bec4538 64bit) > *>* Debian Jessie. Vanilla config. > *>>* I am trying to enable portuguese to say numbers using dptools "Say". > *>>* This is what I did: > *>* - Module mod_say_pt is loaded > *>* - in freeswitch.xml > *>* include lang/pt/pt_PT.xml > *>* - Downloaded sounds from > *>* https://github.com/jpawlowski/freeswitch-sounds-ttsand placed them > *>* into /usr/share/freeswitch/sounds > *>* Directory structure is now like: /pt/tts/google/ascii/16000/... > *>>* CLI> say_string t.wav pt NUMBER pronounced 123 > *>* [ERR] switch_xml.c:3180 Can't find phrases tag > *>>* ..I found out that lang/pt_PT.xml seems to be missing the > *>* tags within the tag (vanilla config). I > *>* fixed that. > *>>* Now: > *>* CLI> say_string t.wav pt NUMBER pronounced 123 > *>* [ERR] switch_ivr.c:3726 Invalid SAY Interface [pt]! > *>>* BTW: English works fine: > *>* CLI> say_string t.wav en NUMBER pronounced 123 > *>* file_string://digits/1.wav!digits/hundred.wav!digits/20.wav!digits/3.wav > *>>* I also tried other languages but have same error "Invalid SAY > *>* Interface". > *>>* What am I missing. Please help. > *> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From hardikitpl at gmail.com Tue Jun 6 11:11:33 2017 From: hardikitpl at gmail.com (Hardik Patel) Date: Tue, 6 Jun 2017 16:41:33 +0530 Subject: [Freeswitch-users] Fax receive issue with t30 codec Message-ID: Hello, I am using opensips as entry point using dispatcher. opensips( 127.0.0.1), i am routing call to freeswitch server (127.0.0.3). Now I am trying to receive fax, my issue is when i try to send fax in softphone(Zoiper) from the log i am seeing that it is sending fax using t30 codec. and i am not receiving the fax at destination, is it because of codec, should it only work with t38 codec? if that is the issue than how am i be able to send the fax using t38 from zoiper? Here i am attaching the fs log with loglevel 9 and sip trace is also enabled. 127.0.0.2 => carrier/provider IP 123456789 => Fax number test at gamil.com => Email Address 127.0.0.4 =>UI IP Pastebin link:https://pastebin.freeswitch.org/view/9ec52715 -------------- next part -------------- An HTML attachment was scrubbed... URL: From deepikay at iiitd.ac.in Tue Jun 6 12:34:51 2017 From: deepikay at iiitd.ac.in (Deepika Yadav) Date: Tue, 6 Jun 2017 18:04:51 +0530 Subject: [Freeswitch-users] Fwd: How to get if of the playing media in a conference In-Reply-To: References: Message-ID: ---------- Forwarded message ---------- From: "Deepika Yadav" Date: Jun 6, 2017 4:29 PM Subject: How to get if of the playing media in a conference To: Cc: Hi, I need to detect the start and stop of media in a conference call through ESL. Event named as "play-file-done" and "play-file" are for detection of any kind of media related activity. However, I could get find a method to get the ID of the media to identify them. e.g. event "play-file-done" is detected multiple times in a conference - if someone is added and also when any playing media is stopped. -- Regards Deepika https://www.iiitd.edu.in/~deepikay/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From wsimon at stratusvideo.com Tue Jun 6 13:21:36 2017 From: wsimon at stratusvideo.com (William Simon) Date: Tue, 6 Jun 2017 13:21:36 +0000 Subject: [Freeswitch-users] X-RTP-STAT or P-RTP-STAT Message-ID: <0362BB4B-BFDA-4BF5-B650-690AB6E35601@stratusvideo.com> Can anyone explain how to add the X/P-RTP-STAT header to the BYE for summarizing RTP statistics at the end of the call? Seems like there's something built in for this, but it's not clear how to use it. https://wiki.freeswitch.org/wiki/Variable_sip_p_rtp_stat "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." -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian at freeswitch.org Tue Jun 6 14:00:06 2017 From: brian at freeswitch.org (Brian West) Date: Tue, 6 Jun 2017 09:00:06 -0500 Subject: [Freeswitch-users] Fax receive issue with t30 codec In-Reply-To: References: Message-ID: You'll have to use 1.6.17 if you ever want any faxing to work in all test cases. /b On Tue, Jun 6, 2017 at 6:11 AM, Hardik Patel wrote: > Hello, > > I am using opensips as entry point using dispatcher. opensips( 127.0.0.1), > i am routing call to freeswitch server (127.0.0.3). > > Now I am trying to receive fax, my issue is when i try to send fax in > softphone(Zoiper) from the log i am seeing that it is sending fax using t30 > codec. and i am not receiving the fax at destination, is it because of > codec, should it only work with t38 codec? if that is the issue than how am > i be able to send the fax using t38 from zoiper? > > Here i am attaching the fs log with loglevel 9 and sip trace is also > enabled. > > 127.0.0.2 => carrier/provider IP > 123456789 => Fax number > test at gamil.com => Email Address > 127.0.0.4 =>UI IP > > Pastebin link:https://pastebin.freeswitch.org/view/9ec52715 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Book a phone call (CST) Allison prompts for FreeSWITCH: *https://www.gofundme.com/allison-prompts-for-freeswitch* Got Bugs? Report them here ! | Reddit: /r/freeswitch *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Tue Jun 6 14:08:19 2017 From: mike at jerris.com (Michael Jerris) Date: Tue, 6 Jun 2017 10:08:19 -0400 Subject: [Freeswitch-users] X-RTP-STAT or P-RTP-STAT In-Reply-To: <0362BB4B-BFDA-4BF5-B650-690AB6E35601@stratusvideo.com> References: <0362BB4B-BFDA-4BF5-B650-690AB6E35601@stratusvideo.com> Message-ID: Those vars are for getting those stats that devices send us in a BYE > On Jun 6, 2017, at 9:21 AM, William Simon wrote: > > Can anyone explain how to add the X/P-RTP-STAT header to the BYE for summarizing RTP statistics at the end of the call? > > Seems like there's something built in for this, but it's not clear how to use it. https://wiki.freeswitch.org/wiki/Variable_sip_p_rtp_stat > > > > > “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.” Sorry, nothing you send to the list is private. it will be distributed. Note, we will NOT remove messages from the list archives if requested (yes, its been requested repeatedly in the past). -------------- next part -------------- An HTML attachment was scrubbed... URL: From wsimon at stratusvideo.com Tue Jun 6 14:25:43 2017 From: wsimon at stratusvideo.com (William Simon) Date: Tue, 6 Jun 2017 14:25:43 +0000 Subject: [Freeswitch-users] X-RTP-STAT or P-RTP-STAT In-Reply-To: References: <0362BB4B-BFDA-4BF5-B650-690AB6E35601@stratusvideo.com> Message-ID: Does FreeSWITCH have the capability to generate the stats? We would like to get a summary RTP quality report from FS's perspective. On Jun 6, 2017, at 10:08 AM, Michael Jerris > wrote: Those vars are for getting those stats that devices send us in a BYE On Jun 6, 2017, at 9:21 AM, William Simon > wrote: Can anyone explain how to add the X/P-RTP-STAT header to the BYE for summarizing RTP statistics at the end of the call? Seems like there's something built in for this, but it's not clear how to use it. https://wiki.freeswitch.org/wiki/Variable_sip_p_rtp_stat “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.” Sorry, nothing you send to the list is private. it will be distributed. Note, we will NOT remove messages from the list archives if requested (yes, its been requested repeatedly in the past). “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 at 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 at 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.” -------------- next part -------------- An HTML attachment was scrubbed... URL: From acheraime at gmail.com Tue Jun 6 14:30:59 2017 From: acheraime at gmail.com (acheraime at gmail.com) Date: Tue, 6 Jun 2017 10:30:59 -0400 Subject: [Freeswitch-users] X-RTP-STAT or P-RTP-STAT In-Reply-To: References: <0362BB4B-BFDA-4BF5-B650-690AB6E35601@stratusvideo.com> Message-ID: <2D844020-558B-4E1E-90CE-3B1B95322BA3@gmail.com> The CDR generated after each call contains a lot of RTP related information include the MOS. Sent from my iPhone > On Jun 6, 2017, at 10:25 AM, William Simon wrote: > > Does FreeSWITCH have the capability to generate the stats? We would like to get a summary RTP quality report from FS's perspective. > >> On Jun 6, 2017, at 10:08 AM, Michael Jerris wrote: >> >> Those vars are for getting those stats that devices send us in a BYE >> >>> On Jun 6, 2017, at 9:21 AM, William Simon wrote: >>> >>> Can anyone explain how to add the X/P-RTP-STAT header to the BYE for summarizing RTP statistics at the end of the call? >>> >>> Seems like there's something built in for this, but it's not clear how to use it. https://wiki.freeswitch.org/wiki/Variable_sip_p_rtp_stat >>> >>> >>> >>> >>> “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.” >> >> Sorry, nothing you send to the list is private. it will be distributed. Note, we will NOT remove messages from the list archives if requested (yes, its been requested repeatedly in the past). >> >> >> >> “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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Tue Jun 6 14:31:50 2017 From: mike at jerris.com (Michael Jerris) Date: Tue, 6 Jun 2017 10:31:50 -0400 Subject: [Freeswitch-users] X-RTP-STAT or P-RTP-STAT In-Reply-To: References: <0362BB4B-BFDA-4BF5-B650-690AB6E35601@stratusvideo.com> Message-ID: We don’t send those in BYE, some stats are recorded in cdr. > On Jun 6, 2017, at 10:25 AM, William Simon wrote: > > Does FreeSWITCH have the capability to generate the stats? We would like to get a summary RTP quality report from FS's perspective. > >> On Jun 6, 2017, at 10:08 AM, Michael Jerris > wrote: >> >> Those vars are for getting those stats that devices send us in a BYE >> >>> On Jun 6, 2017, at 9:21 AM, William Simon > wrote: >>> >>> Can anyone explain how to add the X/P-RTP-STAT header to the BYE for summarizing RTP statistics at the end of the call? >>> >>> Seems like there's something built in for this, but it's not clear how to use it. https://wiki.freeswitch.org/wiki/Variable_sip_p_rtp_stat >>> >>> >>> >>> >>> “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.” >> >> Sorry, nothing you send to the list is private. it will be distributed. Note, we will NOT remove messages from the list archives if requested (yes, its been requested repeatedly in the past). >> >> >> >> “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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From anthony.minessale at gmail.com Tue Jun 6 16:26:50 2017 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 6 Jun 2017 11:26:50 -0500 Subject: [Freeswitch-users] Monitoring call quality using variable_rtp_audio_xxx In-Reply-To: References: Message-ID: flaws are missing or out-of-order packets. If you get subsequent missing packets it counts as 2 flaws. The number of flaws out of the total number of packets is the quality percentage. The quality percentage rounded to a scale of 4.5 is the mos. On Tue, Jun 6, 2017 at 1:59 AM, Jonathan Hunter wrote: > Hi Guys, > > > To be more specific, what is the meaning of this parameter? > > > in_flaw_total > > > I am becoming clear on the other parameters! > > > Thanks > > > Jon > > > ------------------------------ > *From:* Jonathan Hunter > *Sent:* 31 May 2017 19:24 > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* Monitoring call quality using variable_rtp_audio_xxx > > > Hi Guys, > > Sorry for the noise, we are looking to poll calls in progress to grab the > rtp_audio variables when a call is in progress, and we want to understand > if this is a good approach and to that end what values/variables we should > consider and what ranges should we be working with? > > I understand mos and quality percentage but what other values are a good > indicator? > > Again Im clear on jitter and its meaning just want to understand whats > what as not everything understandably is not documented. > > Many thanks > > Jon > > "variable_rtp_audio_recv_pt": "8", > "variable_rtp_audio_in_raw_bytes": "374960", > "variable_rtp_audio_in_media_bytes": "374616", > "variable_rtp_audio_in_packet_count": "2180", > "variable_rtp_audio_in_media_packet_count": "2178", > "variable_rtp_audio_in_skip_packet_count": "92", > "variable_rtp_audio_in_jitter_packet_count": "0", > "variable_rtp_audio_in_dtmf_packet_count": "0", > "variable_rtp_audio_in_cng_packet_count": "0", > "variable_rtp_audio_in_flush_packet_count": "2", > "variable_rtp_audio_in_largest_jb_size": "0", > "variable_rtp_audio_in_jitter_min_variance": "28.57", > "variable_rtp_audio_in_jitter_max_variance": "116.33", > "variable_rtp_audio_in_jitter_loss_rate": "0.02", > "variable_rtp_audio_in_jitter_burst_rate": "0.98", > "variable_rtp_audio_in_mean_interval": "20.39", > "variable_rtp_audio_in_flaw_total": "43", > "variable_rtp_audio_in_quality_percentage": "97.00", > "variable_rtp_audio_in_mos": "4.47", > "variable_rtp_audio_out_raw_bytes": "210012", > "variable_rtp_audio_out_media_bytes": "210012", > "variable_rtp_audio_out_packet_count": "1221", > "variable_rtp_audio_out_media_packet_count": "1221", > "variable_rtp_audio_out_skip_packet_count": "0", > "variable_rtp_audio_out_dtmf_packet_count": "0", > "variable_rtp_audio_out_cng_packet_count": "0", > "variable_rtp_audio_rtcp_packet_count": "0", > "variable_rtp_audio_rtcp_octet_count": "0", > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at 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 ♬ @anthmfs ♬ @FreeSWITCH ♬ ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ * ClueCon Weekly Development Call ☎ sip:888 at conference.freeswitch.org ☎ +19193869900 https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA -------------- next part -------------- An HTML attachment was scrubbed... URL: From mjlopez at smartic.es Tue Jun 6 17:38:30 2017 From: mjlopez at smartic.es (=?iso-8859-1?Q?Miguel_Jes=FAs_L=F3pez_Valverde?=) Date: Tue, 6 Jun 2017 19:38:30 +0200 Subject: [Freeswitch-users] trouble with instalation of mod_fail2ban module in FreeSwitch. In-Reply-To: <52bacbc4-bc14-e9c9-d938-da1cc789f357@madovsky.org> References: <010a01d2daf0$36c6cc50$a45464f0$@smartic.es> <52bacbc4-bc14-e9c9-d938-da1cc789f357@madovsky.org> Message-ID: <0ff201d2deeb$b6fb5090$24f1f1b0$@smartic.es> Thank you very much. Now is running right. De: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] En nombre de Madovsky Enviado el: viernes, 02 de junio de 2017 23:59 Para: freeswitch-users at lists.freeswitch.org Asunto: Re: [Freeswitch-users] trouble with instalation of mod_fail2ban module in FreeSwitch. On 6/1/2017 9:00 AM, Miguel Jesús López Valverde wrote: Hello: I am sending you this email because I am having problems installing the module mod_fail2ban following the recipe offered at https://freeswitch.org/confluence/display/FREESWITCH/mod_fail2ban because when I run make, I get this result: /usr/local/freeswitch/mod/applications/mod_fail2ban$ sudo make Makefile:2: ../../../../build/modmake.rules: No such file or directory make: *** No rule to make target '../../../../build/modmake.rules'. Stop. Do you know in which folder the sentence “git clone” may to be executed?. Do you know if there is anything else to keep in mind that can cause this problem? Thank you very much. Miguel J. Lopez. Libre de virus. www.avast.com did you install fail2ban from your distro? --- El software de antivirus Avast ha analizado este correo electrónico en busca de virus. https://www.avast.com/antivirus -------------- next part -------------- An HTML attachment was scrubbed... URL: From xxxman2008 at 126.com Wed Jun 7 07:42:04 2017 From: xxxman2008 at 126.com (Raymond) Date: Wed, 7 Jun 2017 15:42:04 +0800 (CST) Subject: [Freeswitch-users] question about HA solution In-Reply-To: References: <8A13E0EC-FA13-4267-80F6-CE1A8E8360CF@jerris.com> Message-ID: <55e4ea34.7356.15c8180dc29.Coremail.xxxman2008@126.com> Ok, Denys , I understand you have an "Master - Slave" situation. So ,you must move the call back when Master server come back. I'm sorry ,no help for u ,but ,maybe ,use an " master - master - master -...." architecture is the fastest way to resolve the problem. :-)) . The key of your question is "auto-clear-sql" option not functioning normally . plz double check your config ,and make sure , all servers have same option. Raymond 在 2017-06-06 01:50:18，"Denys Pozniak" 写道： Yes, correct. But when you restart FS on slave, it will erase database. And option auto-clear-sql=false not working for me. On Jun 5, 2017 6:32 PM, "Michael Jerris" wrote: recovered calls will get new entries in the table. -------------- next part -------------- An HTML attachment was scrubbed... URL: From hunterj91 at hotmail.com Wed Jun 7 08:50:09 2017 From: hunterj91 at hotmail.com (Jonathan Hunter) Date: Wed, 7 Jun 2017 08:50:09 +0000 Subject: [Freeswitch-users] Monitoring call quality using variable_rtp_audio_xxx In-Reply-To: References: , Message-ID: Perfect thank you! ________________________________ From: FreeSWITCH-users on behalf of Anthony Minessale Sent: 06 June 2017 16:26 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Monitoring call quality using variable_rtp_audio_xxx flaws are missing or out-of-order packets. If you get subsequent missing packets it counts as 2 flaws. The number of flaws out of the total number of packets is the quality percentage. The quality percentage rounded to a scale of 4.5 is the mos. On Tue, Jun 6, 2017 at 1:59 AM, Jonathan Hunter > wrote: Hi Guys, To be more specific, what is the meaning of this parameter? in_flaw_total I am becoming clear on the other parameters! Thanks Jon ________________________________ From: Jonathan Hunter > Sent: 31 May 2017 19:24 To: freeswitch-users at lists.freeswitch.org Subject: Monitoring call quality using variable_rtp_audio_xxx Hi Guys, Sorry for the noise, we are looking to poll calls in progress to grab the rtp_audio variables when a call is in progress, and we want to understand if this is a good approach and to that end what values/variables we should consider and what ranges should we be working with? I understand mos and quality percentage but what other values are a good indicator? Again Im clear on jitter and its meaning just want to understand whats what as not everything understandably is not documented. Many thanks Jon "variable_rtp_audio_recv_pt": "8", "variable_rtp_audio_in_raw_bytes": "374960", "variable_rtp_audio_in_media_bytes": "374616", "variable_rtp_audio_in_packet_count": "2180", "variable_rtp_audio_in_media_packet_count": "2178", "variable_rtp_audio_in_skip_packet_count": "92", "variable_rtp_audio_in_jitter_packet_count": "0", "variable_rtp_audio_in_dtmf_packet_count": "0", "variable_rtp_audio_in_cng_packet_count": "0", "variable_rtp_audio_in_flush_packet_count": "2", "variable_rtp_audio_in_largest_jb_size": "0", "variable_rtp_audio_in_jitter_min_variance": "28.57", "variable_rtp_audio_in_jitter_max_variance": "116.33", "variable_rtp_audio_in_jitter_loss_rate": "0.02", "variable_rtp_audio_in_jitter_burst_rate": "0.98", "variable_rtp_audio_in_mean_interval": "20.39", "variable_rtp_audio_in_flaw_total": "43", "variable_rtp_audio_in_quality_percentage": "97.00", "variable_rtp_audio_in_mos": "4.47", "variable_rtp_audio_out_raw_bytes": "210012", "variable_rtp_audio_out_media_bytes": "210012", "variable_rtp_audio_out_packet_count": "1221", "variable_rtp_audio_out_media_packet_count": "1221", "variable_rtp_audio_out_skip_packet_count": "0", "variable_rtp_audio_out_dtmf_packet_count": "0", "variable_rtp_audio_out_cng_packet_count": "0", "variable_rtp_audio_rtcp_packet_count": "0", "variable_rtp_audio_rtcp_octet_count": "0", _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at 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 at 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 ♬ @anthmfs ♬ @FreeSWITCH ♬ ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ http://freeswitch.org/g+ ClueCon Weekly Development Call ☎ sip:888 at conference.freeswitch.org ☎ +19193869900 https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA -------------- next part -------------- An HTML attachment was scrubbed... URL: From boesemar at gmail.com Wed Jun 7 09:22:45 2017 From: boesemar at gmail.com (boesemar .) Date: Wed, 7 Jun 2017 10:22:45 +0100 Subject: [Freeswitch-users] Enabling new languages for Say In-Reply-To: References: Message-ID: Fixed! It seems that just say_string on the CLI is only available for en and ru. Also mentioned here: https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_say Application 'say' from the dialplan is in fact working... all good :) Martin On Mon, Jun 5, 2017 at 10:53 AM, boesemar . wrote: > Hi Brian, > > mod_say_pt is definitely loaded: > > > module_exists mod_say_pt > true > > Can you think of any other cause for the "Invalid SAY Interface [pt]" ? > > Martin > > > PS: sorry for breaking the thread - problem with email > > > > Brian West brian at freeswitch.org > Fri Jun 2 14:51:25 UTC 2017 > > I would guess mod_say_pt is not loaded. > > On Fri, Jun 2, 2017 at 4:54 AM, Martin Boese > wrote: > > >* Hi! > *>>* FreeSWITCH Version 1.6.15-32-bec4538~64bit (-32-bec4538 64bit) > *>* Debian Jessie. Vanilla config. > *>>* I am trying to enable portuguese to say numbers using dptools "Say". > *>>* This is what I did: > *>* - Module mod_say_pt is loaded > *>* - in freeswitch.xml > *>* include lang/pt/pt_PT.xml > *>* - Downloaded sounds from > *>* https://github.com/jpawlowski/freeswitch-sounds-ttsand placed them > *>* into /usr/share/freeswitch/sounds > *>* Directory structure is now like: /pt/tts/google/ascii/16000/... > *>>* CLI> say_string t.wav pt NUMBER pronounced 123 > *>* [ERR] switch_xml.c:3180 Can't find phrases tag > *>>* ..I found out that lang/pt_PT.xml seems to be missing the > *>* tags within the tag (vanilla config). I > *>* fixed that. > *>>* Now: > *>* CLI> say_string t.wav pt NUMBER pronounced 123 > *>* [ERR] switch_ivr.c:3726 Invalid SAY Interface [pt]! > *>>* BTW: English works fine: > *>* CLI> say_string t.wav en NUMBER pronounced 123 > *>* file_string://digits/1.wav!digits/hundred.wav!digits/20.wav!digits/3.wav > *>>* I also tried other languages but have same error "Invalid SAY > *>* Interface". > *>>* What am I missing. Please help. > *>>* Thanks, > *>* Martin > *>>* _________________________________________________________________________ > *>* Professional FreeSWITCH Consulting Services: > *>* consulting at 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 at lists.freeswitch.org > *>* http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > *>* UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > *>* http://www.freeswitch.org > * > > > > -- > > *Brian West*brian at freeswitch.org > > *Twitter: @FreeSWITCH , @briankwest* > http://www.freeswitchbook.comhttp://www.freeswitchcookbook.com > > Book a phone call (CST) > > Allison prompts for FreeSWITCH: > > *https://www.gofundme.com/allison-prompts-for-freeswitch* > > > Got Bugs? Report them here ! | Reddit: > /r/freeswitch > > *T:*+19184209001 <%28918%29%20420-9001> | *F:*+19184209002 <%28918%29%20420-9002> | *M:*+1918424WEST (9378) > *Skype:*briankwest > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > > ------------------------------ > -------------- next part -------------- An HTML attachment was scrubbed... URL: From Alexander.Haugg at c4b.de Wed Jun 7 10:49:30 2017 From: Alexander.Haugg at c4b.de (Alexander Haugg) Date: Wed, 7 Jun 2017 10:49:30 +0000 Subject: [Freeswitch-users] Turn configuration mod_sofia Message-ID: <26962017ab7e4ba88fbd2db5a551a34d@c4b.de> Hi, my question: how can I configure an ICE server (turn) in the sip profile? My think for this id case I need the possibility to configure the turn server with credentials in the sip profile like the stun configuration too. - STUN configuration "" -> that's working! - ICE configuration could be? "" Is there a possibility? Thanks a lot Von: Alexander Haugg Gesendet: Donnerstag, 1. Juni 2017 15:24 An: 'freeswitch-users at lists.freeswitch.org' Betreff: WG: Turn configuration mod_sofia Hi Brian, we use the Freeswitch as "Man in the Middle" for WebRTC. The WebRTC Clients (ICE Link Stack from Frozen Mountain) registered via SIP on the Freeswitch. The SIP signalling is a special szenario and works successfully in several LAN WAN setups. But now we need srflx (that's fine with teh STUN configuration) an relay candidates in the SDP that's ganerated by the Freeswitch. My think is, for this case i need the possibility to configure the turn server with credentials in the sip profile like the stun configuration too. - STUN configuration "" - ICE configuration could be? "" Thanks a lot >From brian at freeswitch.org Tue May 30 13:50:32 2017 From: brian at freeswitch.org (Brian West) Date: Tue, 30 May 2017 08:50:32 -0500 Subject: [Freeswitch-users] Turn configuration mod_sofia In-Reply-To: > References: > Message-ID: > There is little reason to use TURN when speaking to FreeSWITCH, What issue are you trying to solve? Von: Alexander Haugg Gesendet: Montag, 29. Mai 2017 18:25 An: 'freeswitch-users at lists.freeswitch.org' > Betreff: Turn configuration mod_sofia Hi all, the stun configuration for the mod_sofia profile is very easy "" but what is to do, if i need a relay candidate in the sdp? How can i set the turn address and the login credentials? Thanks a lot -------------- next part -------------- An HTML attachment was scrubbed... URL: From khorsmann at gmail.com Wed Jun 7 11:47:03 2017 From: khorsmann at gmail.com (Karsten Horsmann) Date: Wed, 7 Jun 2017 13:47:03 +0200 Subject: [Freeswitch-users] Multiple FreeSWITCH servers behind kamailio-websocket Message-ID: Hello List, is there any howto about webrtc loadbalance in combination with kamailio and FreeSWITCH? I want to share one WSS address/endpoint to multiple FreeSWITCH backends. Or is there any other best practice? My callflow is mostly that my internal SIP Servers called my registered webrtc clients. Would be nice to get some input. -- Kind Regards *Karsten Horsmann* -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Wed Jun 7 15:35:25 2017 From: mike at jerris.com (Michael Jerris) Date: Wed, 7 Jun 2017 11:35:25 -0400 Subject: [Freeswitch-users] question about HA solution In-Reply-To: References: <8A13E0EC-FA13-4267-80F6-CE1A8E8360CF@jerris.com> Message-ID: That param should keep it from doing so, if its not you are not setting it somehow or something else is wiping the db. > On Jun 5, 2017, at 1:50 PM, Denys Pozniak wrote: > > Yes, correct. But when you restart FS on slave, it will erase database. And option auto-clear-sql=false not working for me. > > On Jun 5, 2017 6:32 PM, "Michael Jerris" > wrote: > recovered calls will get new entries in the table. > >> On Jun 5, 2017, at 7:41 AM, Denys Pozniak > wrote: >> >> Hello! >> >> Thank you Raymond about your explanation, but I dont agree with some point: >> If it really need an answer about your question -- "if it is possible to move calls back". I think it's unnecessary. - in my case I have two not equal servers, so I need to have only one as a master. >> If switchover happens I need to have ability to restore master back. >> >> Thank you Luis for your link, you can do simple test to understand what I am talking about: do call -> check on master and slave #show channels -> restart FS on slave -> check on master #show channels. In my case I dont see any active calls after this, so restoring back is not possible. >> >> >> >> On 3 June 2017 at 22:16, Luis Daniel Lucio Quiroz > wrote: >> You may want to read this article. >> >> http://inside-out.xyz/technology/how-to-configure-freeswitch-for-ha.html >> >> Le 31 mai 2017 6:29 PM, "Denys Pozniak" > a écrit : >> Hello! >> >> I built FS HA solution based on keepalived and mysql master-master. >> It works ok generally, but as I understand FS after restarting cleaning own database. >> >> So when node1 fails calls jump to node2, after script restarts node1 it is not possible to move calls back. >> >> Tried options in switch.conf.xml, but no luck: >> >> >> >> >> Is there is a way to solve this? >> -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Wed Jun 7 15:41:53 2017 From: mike at jerris.com (Michael Jerris) Date: Wed, 7 Jun 2017 11:41:53 -0400 Subject: [Freeswitch-users] Enabling new languages for Say In-Reply-To: References: Message-ID: Thats weird, its in more than en and ru, its also in es_ar, he, pl, and sv. Not sure why whoever wrote the other language mods never added that part. If anyone wants to toss me a pull request on fixing this, I’d be happy to look at it. > On Jun 7, 2017, at 5:22 AM, boesemar . wrote: > > Fixed! > > It seems that just say_string on the CLI is only available for en and ru. Also mentioned here: > https://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_say > > Application 'say' from the dialplan is in fact working... all good :) > > Martin > > On Mon, Jun 5, 2017 at 10:53 AM, boesemar . > wrote: > Hi Brian, > > mod_say_pt is definitely loaded: > > > module_exists mod_say_pt > true > > Can you think of any other cause for the "Invalid SAY Interface [pt]" ? > Martin > > PS: sorry for breaking the thread - problem with email > > > Brian West brian at freeswitch.org > Fri Jun 2 14:51:25 UTC 2017 > > I would guess mod_say_pt is not loaded. > > On Fri, Jun 2, 2017 at 4:54 AM, Martin Boese > wrote: > > > Hi! > > > > FreeSWITCH Version 1.6.15-32-bec4538~64bit (-32-bec4538 64bit) > > Debian Jessie. Vanilla config. > > > > I am trying to enable portuguese to say numbers using dptools "Say". > > > > This is what I did: > > - Module mod_say_pt is loaded > > - in freeswitch.xml > > include lang/pt/pt_PT.xml > > - Downloaded sounds from > > https://github.com/jpawlowski/freeswitch-sounds-ttsand placed them > > into /usr/share/freeswitch/sounds > > Directory structure is now like: /pt/tts/google/ascii/16000/... > > > > CLI> say_string t.wav pt NUMBER pronounced 123 > > [ERR] switch_xml.c:3180 Can't find phrases tag > > > > ..I found out that lang/pt_PT.xml seems to be missing the > > tags within the tag (vanilla config). I > > fixed that. > > > > Now: > > CLI> say_string t.wav pt NUMBER pronounced 123 > > [ERR] switch_ivr.c:3726 Invalid SAY Interface [pt]! > > > > BTW: English works fine: > > CLI> say_string t.wav en NUMBER pronounced 123 > > file_string://digits/1.wav!digits/hundred.wav!digits/20.wav!digits/3.wav <> > > > > I also tried other languages but have same error "Invalid SAY > > Interface". > > > > What am I missing. Please help. > > > > Thanks, > > Martin > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at 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 at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > -- > > *Brian West* > brian at freeswitch.org > > *Twitter: @FreeSWITCH , @briankwest* > > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > Book a phone call (CST) > > > Allison prompts for FreeSWITCH: > > *https://www.gofundme.com/allison-prompts-for-freeswitch* > > > > Got Bugs? Report them here >! | Reddit: > /r/freeswitch > > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *Skype:*briankwest > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From mase2hot at gmail.com Wed Jun 7 15:49:39 2017 From: mase2hot at gmail.com (Jason Bedward) Date: Wed, 7 Jun 2017 16:49:39 +0100 Subject: [Freeswitch-users] Re-invite Glare issue - FS ending call Message-ID: Hi, I have an issue with re-invites on some calls. I'm using 1.6.15 and these calls are using bypass media. I have Kamailio as inbound SBC but FS connecting directly to my provider for B leg. - After 5 minutes plus on some calls my provider sends re-invite on A leg - FS then sends this re-invite to the B leg - At the same time the B leg sends a Re-invite - FS replys 491 - B leg provider replys 100, then 500 (with retry in the 500) - FS send ACK and then BYE Not sure what setting I need to change or can change infact to either retry the invite in accordance with the 500 retry request. Or something else to stop the call ending... [image: Inline image 1] Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image.png Type: image/png Size: 45647 bytes Desc: not available URL: From mike at jerris.com Wed Jun 7 15:53:00 2017 From: mike at jerris.com (Michael Jerris) Date: Wed, 7 Jun 2017 11:53:00 -0400 Subject: [Freeswitch-users] Turn configuration mod_sofia In-Reply-To: <26962017ab7e4ba88fbd2db5a551a34d@c4b.de> References: <26962017ab7e4ba88fbd2db5a551a34d@c4b.de> Message-ID: you would configure a TURN server on the client. we will relay through a turn server when supplied by the client, but as we are a server stack, we have no support for a turn server on the server side, as Brain said, there is little reason for it on the server side. > On Jun 7, 2017, at 6:49 AM, Alexander Haugg wrote: > > Hi, > > my question: > how can I configure an ICE server (turn) in the sip profile? > > My think for this id case I need the possibility to configure the turn server with credentials in the sip profile like the stun configuration too. > - STUN configuration „“ -> that‘s working! > - ICE configuration could be? „“ > > Is there a possibility? > > Thanks a lot > > Von: Alexander Haugg > Gesendet: Donnerstag, 1. Juni 2017 15:24 > An: 'freeswitch-users at lists.freeswitch.org ' > > Betreff: WG: Turn configuration mod_sofia > > Hi Brian, > > we use the Freeswitch as „Man in the Middle“ for WebRTC. > > The WebRTC Clients (ICE Link Stack from Frozen Mountain) registered via SIP on the Freeswitch. > The SIP signalling is a special szenario and works successfully in several LAN WAN setups. > But now we need srflx (that’s fine with teh STUN configuration) an relay candidates in the SDP that’s ganerated by the Freeswitch. > > My think is, for this case i need the possibility to configure the turn server with credentials in the sip profile like the stun configuration too. > - STUN configuration „“ > - ICE configuration could be? „“ > > Thanks a lot > > > From brian at freeswitch.org Tue May 30 13:50:32 2017 > From: brian at freeswitch.org (Brian West) > Date: Tue, 30 May 2017 08:50:32 -0500 > Subject: [Freeswitch-users] Turn configuration mod_sofia > In-Reply-To: > > References: > > Message-ID: > > > There is little reason to use TURN when speaking to FreeSWITCH, What issue > are you trying to solve? > > > Von: Alexander Haugg > Gesendet: Montag, 29. Mai 2017 18:25 > An: 'freeswitch-users at lists.freeswitch.org ' > > Betreff: Turn configuration mod_sofia > > Hi all, > > the stun configuration for the mod_sofia profile is very easy „“ > but what is to do, if i need a relay candidate in the sdp? How can i set the turn address and the login credentials? > > Thanks a lot > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Wed Jun 7 15:57:18 2017 From: mike at jerris.com (Michael Jerris) Date: Wed, 7 Jun 2017 11:57:18 -0400 Subject: [Freeswitch-users] Re-invite Glare issue - FS ending call In-Reply-To: References: Message-ID: <72A53E3A-47FC-498E-AF90-2B39448DE52A@jerris.com> why are they both sending a re-invite at the same time? This is called glare, we seem to be handling it properly, the provider seems not to be. The easiest fix here is probably to figure out why we are both sending re-invite at the same time, maybe session timers and the provider is broken and trying to send it when we said we would. > On Jun 7, 2017, at 11:49 AM, Jason Bedward wrote: > > Hi, > > I have an issue with re-invites on some calls. I'm using 1.6.15 and these calls are using bypass media. I have Kamailio as inbound SBC but FS connecting directly to my provider for B leg. > > After 5 minutes plus on some calls my provider sends re-invite on A leg > FS then sends this re-invite to the B leg > At the same time the B leg sends a Re-invite > FS replys 491 > B leg provider replys 100, then 500 (with retry in the 500) > FS send ACK and then BYE > Not sure what setting I need to change or can change infact to either retry the invite in accordance with the 500 retry request. Or something else to stop the call ending... > > > > Thanks > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From grcamauer at gmail.com Wed Jun 7 20:17:33 2017 From: grcamauer at gmail.com (Guillermo Ruiz Camauer) Date: Wed, 7 Jun 2017 17:17:33 -0300 Subject: [Freeswitch-users] Open Source SIP\WebRTC clients compatible with FreeSWITCH In-Reply-To: <10cd01d236a8$425136b0$c6f3a410$@freeswitch.org> References: <6FD2F8B5BB72834E9939AEDF9FB802A901E860B047@mbx-01.sysconfig.co.uk> <10c301d236a7$1ea0f8f0$5be2ead0$@freeswitch.org> <10cd01d236a8$425136b0$c6f3a410$@freeswitch.org> Message-ID: Apple announces WebRTC support in iOS11 / Safari: https://apple.slashdot.org/story/17/06/07/1958242/apple-announces-support-for-webrtc-in-safari-11 On Fri, Nov 4, 2016 at 11:32 AM, Ken Rice wrote: > No one supports Native WebRTC on iOS at this time except for people using > their own private SDKs that they are not allowing to get out there… > > > > Apple does not have webRTC in webkit (Safari) or iOS at this time. Chrome > on iOS is not even really Chrome, its just a wrapper around the WebKIT APIs > and is effectively just safari with a few extra functions and built to look > like chrome. > > > > > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Chandramouli > P > *Sent:* Friday, November 4, 2016 9:29 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Open Source SIP\WebRTC clients > compatible with FreeSWITCH > > > > Hello Ken, > > > > We worked with Google native WebRTC on Firefox, Chrome, and Opera browsers > on Windows OS. I recently noticed that Google has added the WebRTC support > for Android, and iOS platforms (webrtc.org). Now, we are planning to > develop video calling module using Google native WebRTC on these new > platforms. Can anybody give me the information about my below queries: > > > > 1) Does Google native WebRTC supports Apple iOS platform (native mobile > app)? > > 2) Does Google native WebRTC supports Apple OS X platform? > > 3) Is it possible to develop video calling module using native WebRTC on > Safari, and Chrome browsers on Apple OS X platform? > > 4) Does Google native WebRTC supports Android platform (native mobile app)? > > 5) If it supports, I could not find any documentation for Apple iOS, Apple > OS X, and Android platforms specifically. Could you please send some > referral links? > > 6) I could not able to find the referral examples also for Apple iOS, > Apple OS X, and Android platforms specifically. Could you please send some > referral links? > > > > Please do needful. > > > > Thank you, > > Chandramouli. > > > > > > On Fri, Nov 4, 2016 at 7:54 PM, Ken Rice wrote: > > Verto Works on pretty much any platform that has native webrtc support > now... unfortunately things like iOS and don’t have native iOS support yet… > > > > If you are looking to build something you might contact > consulting at freeswitch.org and see if you can work with the FSS Team to > develop something > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Shaun Stokes > *Sent:* Friday, November 4, 2016 8:40 AM > *To:* 'FreeSWITCH Users Help' > *Subject:* [Freeswitch-users] Open Source SIP\WebRTC clients compatible > with FreeSWITCH > > > > Hi All, > > > > Does anyone have any recommendations on a good open source SIP\WebRTC > client which works on multiple platforms (Windows, Mac, Linux, Mobiles) to > provide presence, voice, video, instant messaging, screen sharing and file > sharing? This must be capable of integrating with FreeSWITCH for voice and > video (presence via FreeSWITCH would be an advantage). > > > > Many Thanks, > > Shaun > > Shaun Stokes - Infrastructure Analyst > > T : > > 01453 700713 > > E : > > shaun.stokes at itec-support.co.uk > > W : > > www.itec-support.co.uk > > Registered Address :- ITEC Support, Suite 2 Prospect House, Bath Road, > Stroud, Gloucestershire GL5 3QF > Company No. 06908001 > > > CONFIDENTIALITY NOTICE > This communication and the information it contains are intended for the > person or organisation to which it is addressed. Its contents are > confidential and may be protected in law. Unauthorised use, copying or > disclosure of any of it may be unlawful. If you are not the intended > recipient, please contact us immediately. > The contents of any attachments in this e-mail may contain software > viruses, which could damage your own computer system. While ITEC Support > has taken every reasonable precaution to minimise this risk, we cannot > accept liability for any damage which you sustain as a result of software > viruses. You should carry out your own virus checking procedure before > opening any attachment. > > > ______________________________________________________________________ > This message has been checked for all known viruses by MessageLabs Virus > Scanning Service. > ______________________________________________________________________ > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Guillermo Ruiz Camauer -------------- next part -------------- An HTML attachment was scrubbed... URL: From savolainen at erinaco.ru Wed Jun 7 18:16:45 2017 From: savolainen at erinaco.ru (Dmitri Savolainen) Date: Wed, 7 Jun 2017 21:16:45 +0300 Subject: [Freeswitch-users] [SR-Users] Multiple FreeSWITCH servers behind kamailio-websocket In-Reply-To: References: Message-ID: webrtc kamailio for example here https://github.com/havfo/WEBRTC-to-SIP By the way rtpengine is not mandatory with FreeSwitch. It is possible to use a set of FS(1.6) and balancing by dispatcher module 2017-06-07 14:47 GMT+03:00 Karsten Horsmann : > Hello List, > > > is there any howto about webrtc loadbalance in combination with kamailio > and FreeSWITCH? > > I want to share one WSS address/endpoint to multiple FreeSWITCH backends. > Or is there any other best practice? > > My callflow is mostly that my internal SIP Servers called my registered > webrtc clients. > > Would be nice to get some input. > > -- > Kind Regards > *Karsten Horsmann* > > _______________________________________________ > Kamailio (SER) - Users Mailing List > sr-users at lists.kamailio.org > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > -- Savolainen Dmitri -------------- next part -------------- An HTML attachment was scrubbed... URL: From anthony.minessale at gmail.com Thu Jun 8 00:44:32 2017 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 08 Jun 2017 00:44:32 +0000 Subject: [Freeswitch-users] question about HA solution In-Reply-To: References: <8A13E0EC-FA13-4267-80F6-CE1A8E8360CF@jerris.com> Message-ID: Which tables in particular? The one related to call recovery is the only one that matters and it never auto-clears On Wed, Jun 7, 2017 at 10:36 AM Michael Jerris wrote: > That param should keep it from doing so, if its not you are not setting it > somehow or something else is wiping the db. > > On Jun 5, 2017, at 1:50 PM, Denys Pozniak > wrote: > > Yes, correct. But when you restart FS on slave, it will erase database. > And option auto-clear-sql=false not working for me. > > On Jun 5, 2017 6:32 PM, "Michael Jerris" wrote: > > recovered calls will get new entries in the table. > > On Jun 5, 2017, at 7:41 AM, Denys Pozniak > wrote: > > Hello! > > Thank you *Raymond* about your explanation, but I dont agree with some > point: > *If it really need an answer about your question -- "if it is possible to > move calls back". I think it's unnecessary.* - in my case I have two > not equal servers, so I need to have only one as a master. > If switchover happens I need to have ability to restore master back. > > Thank you *Luis* for your link, you can do simple test to understand what > I am talking about: do call -> check on master and slave #show channels -> > restart FS on slave -> check on master #show channels. In my case I dont > see any active calls after this, so restoring back is not possible. > > > > On 3 June 2017 at 22:16, Luis Daniel Lucio Quiroz < > luis.daniel.lucio at gmail.com> wrote: > >> You may want to read this article. >> >> http://inside-out.xyz/technology/how-to-configure-freeswitch-for-ha.html >> >> Le 31 mai 2017 6:29 PM, "Denys Pozniak" a >> écrit : >> >> Hello! >> >> I built FS HA solution based on keepalived and mysql master-master. >> It works ok generally, but as I understand FS after restarting cleaning >> own database. >> >> So when node1 fails calls jump to node2, after script restarts node1 it >> is not possible to move calls back. >> >> Tried options in switch.conf.xml, but no luck: >> >> >> >> >> Is there is a way to solve this? >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at 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 ♬ @anthmfs ♬ @FreeSWITCH ♬ ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ * ClueCon Weekly Development Call ☎ sip:888 at conference.freeswitch.org ☎ +19193869900 https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA -------------- next part -------------- An HTML attachment was scrubbed... URL: From rmundkowsky at ets.org Wed Jun 7 21:50:31 2017 From: rmundkowsky at ets.org (Mundkowsky, Robert) Date: Wed, 7 Jun 2017 21:50:31 +0000 Subject: [Freeswitch-users] Open Source SIP\WebRTC clients compatible with FreeSWITCH In-Reply-To: References: <6FD2F8B5BB72834E9939AEDF9FB802A901E860B047@mbx-01.sysconfig.co.uk> <10c301d236a7$1ea0f8f0$5be2ead0$@freeswitch.org> <10cd01d236a8$425136b0$c6f3a410$@freeswitch.org> Message-ID: Cool, not if they just support getUserMedia! Robert Mundkowsky From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Guillermo Ruiz Camauer Sent: Wednesday, June 7, 2017 4:18 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Open Source SIP\WebRTC clients compatible with FreeSWITCH Apple announces WebRTC support in iOS11 / Safari: https://apple.slashdot.org/story/17/06/07/1958242/apple-announces-support-for-webrtc-in-safari-11 On Fri, Nov 4, 2016 at 11:32 AM, Ken Rice > wrote: No one supports Native WebRTC on iOS at this time except for people using their own private SDKs that they are not allowing to get out there… Apple does not have webRTC in webkit (Safari) or iOS at this time. Chrome on iOS is not even really Chrome, its just a wrapper around the WebKIT APIs and is effectively just safari with a few extra functions and built to look like chrome. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Chandramouli P Sent: Friday, November 4, 2016 9:29 AM To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Open Source SIP\WebRTC clients compatible with FreeSWITCH Hello Ken, We worked with Google native WebRTC on Firefox, Chrome, and Opera browsers on Windows OS. I recently noticed that Google has added the WebRTC support for Android, and iOS platforms (webrtc.org). Now, we are planning to develop video calling module using Google native WebRTC on these new platforms. Can anybody give me the information about my below queries: 1) Does Google native WebRTC supports Apple iOS platform (native mobile app)? 2) Does Google native WebRTC supports Apple OS X platform? 3) Is it possible to develop video calling module using native WebRTC on Safari, and Chrome browsers on Apple OS X platform? 4) Does Google native WebRTC supports Android platform (native mobile app)? 5) If it supports, I could not find any documentation for Apple iOS, Apple OS X, and Android platforms specifically. Could you please send some referral links? 6) I could not able to find the referral examples also for Apple iOS, Apple OS X, and Android platforms specifically. Could you please send some referral links? Please do needful. Thank you, Chandramouli. On Fri, Nov 4, 2016 at 7:54 PM, Ken Rice > wrote: Verto Works on pretty much any platform that has native webrtc support now... unfortunately things like iOS and don’t have native iOS support yet… If you are looking to build something you might contact consulting at freeswitch.org and see if you can work with the FSS Team to develop something From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Shaun Stokes Sent: Friday, November 4, 2016 8:40 AM To: 'FreeSWITCH Users Help' > Subject: [Freeswitch-users] Open Source SIP\WebRTC clients compatible with FreeSWITCH Hi All, Does anyone have any recommendations on a good open source SIP\WebRTC client which works on multiple platforms (Windows, Mac, Linux, Mobiles) to provide presence, voice, video, instant messaging, screen sharing and file sharing? This must be capable of integrating with FreeSWITCH for voice and video (presence via FreeSWITCH would be an advantage). Many Thanks, Shaun [http://www.itec-support.co.uk/wp-content/uploads/2016/07/email_logo.jpg] Shaun Stokes - Infrastructure Analyst T : 01453 700713 E : shaun.stokes at itec-support.co.uk W : www.itec-support.co.uk Registered Address :- ITEC Support, Suite 2 Prospect House, Bath Road, Stroud, Gloucestershire GL5 3QF Company No. 06908001 CONFIDENTIALITY NOTICE This communication and the information it contains are intended for the person or organisation to which it is addressed. Its contents are confidential and may be protected in law. Unauthorised use, copying or disclosure of any of it may be unlawful. If you are not the intended recipient, please contact us immediately. The contents of any attachments in this e-mail may contain software viruses, which could damage your own computer system. While ITEC Support has taken every reasonable precaution to minimise this risk, we cannot accept liability for any damage which you sustain as a result of software viruses. You should carry out your own virus checking procedure before opening any attachment. ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at 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 at 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 at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Guillermo Ruiz Camauer ________________________________ This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited. Thank you for your compliance. ________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL: From Alexander.Haugg at c4b.de Thu Jun 8 06:34:23 2017 From: Alexander.Haugg at c4b.de (Alexander Haugg) Date: Thu, 8 Jun 2017 06:34:23 +0000 Subject: [Freeswitch-users] Turn configuration mod_sofia In-Reply-To: References: <26962017ab7e4ba88fbd2db5a551a34d@c4b.de> Message-ID: <0b9ea6953aef4324b9c478933e5235aa@c4b.de> Hi, now i know what you mean. Thanks for the right direction! „Mastering Freeswitch“ is a pretty nice book! Von: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Michael Jerris Gesendet: Mittwoch, 7. Juni 2017 17:53 An: FreeSWITCH Users Help Betreff: Re: [Freeswitch-users] Turn configuration mod_sofia you would configure a TURN server on the client. we will relay through a turn server when supplied by the client, but as we are a server stack, we have no support for a turn server on the server side, as Brain said, there is little reason for it on the server side. On Jun 7, 2017, at 6:49 AM, Alexander Haugg > wrote: Hi, my question: how can I configure an ICE server (turn) in the sip profile? My think for this id case I need the possibility to configure the turn server with credentials in the sip profile like the stun configuration too. - STUN configuration „“ -> that‘s working! - ICE configuration could be? „“ Is there a possibility? Thanks a lot Von: Alexander Haugg Gesendet: Donnerstag, 1. Juni 2017 15:24 An: 'freeswitch-users at lists.freeswitch.org' > Betreff: WG: Turn configuration mod_sofia Hi Brian, we use the Freeswitch as „Man in the Middle“ for WebRTC. The WebRTC Clients (ICE Link Stack from Frozen Mountain) registered via SIP on the Freeswitch. The SIP signalling is a special szenario and works successfully in several LAN WAN setups. But now we need srflx (that’s fine with teh STUN configuration) an relay candidates in the SDP that’s ganerated by the Freeswitch. My think is, for this case i need the possibility to configure the turn server with credentials in the sip profile like the stun configuration too. - STUN configuration „“ - ICE configuration could be? „“ Thanks a lot From brian at freeswitch.org Tue May 30 13:50:32 2017 From: brian at freeswitch.org (Brian West) Date: Tue, 30 May 2017 08:50:32 -0500 Subject: [Freeswitch-users] Turn configuration mod_sofia In-Reply-To: > References: > Message-ID: > There is little reason to use TURN when speaking to FreeSWITCH, What issue are you trying to solve? Von: Alexander Haugg Gesendet: Montag, 29. Mai 2017 18:25 An: 'freeswitch-users at lists.freeswitch.org' > Betreff: Turn configuration mod_sofia Hi all, the stun configuration for the mod_sofia profile is very easy „“ but what is to do, if i need a relay candidate in the sdp? How can i set the turn address and the login credentials? Thanks a lot _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From mase2hot at gmail.com Thu Jun 8 07:14:00 2017 From: mase2hot at gmail.com (Jason Bedward) Date: Thu, 8 Jun 2017 08:14:00 +0100 Subject: [Freeswitch-users] Re-invite Glare issue - FS ending call In-Reply-To: <72A53E3A-47FC-498E-AF90-2B39448DE52A@jerris.com> References: <72A53E3A-47FC-498E-AF90-2B39448DE52A@jerris.com> Message-ID: My provider is sending the reinvite on the A leg FS the sends this on to the B leg. At same time the B leg provider sends a reinvite. I dont believe FS is sending the reinvite on its own accord. Its only because of the A leg and that we are using bypass media. So I dont think session timers will make a difference. FYI A leg and B leg provider are the same company although different servers. On 7 Jun 2017 16:57, "Michael Jerris" wrote: > why are they both sending a re-invite at the same time? This is called > glare, we seem to be handling it properly, the provider seems not to be. > The easiest fix here is probably to figure out why we are both sending > re-invite at the same time, maybe session timers and the provider is broken > and trying to send it when we said we would. > > On Jun 7, 2017, at 11:49 AM, Jason Bedward wrote: > > Hi, > > I have an issue with re-invites on some calls. I'm using 1.6.15 and these > calls are using bypass media. I have Kamailio as inbound SBC but FS > connecting directly to my provider for B leg. > > > - After 5 minutes plus on some calls my provider sends re-invite on A > leg > - FS then sends this re-invite to the B leg > - At the same time the B leg sends a Re-invite > - FS replys 491 > - B leg provider replys 100, then 500 (with retry in the 500) > - FS send ACK and then BYE > > Not sure what setting I need to change or can change infact to either > retry the invite in accordance with the 500 retry request. Or something > else to stop the call ending... > > > > Thanks > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From s.safarov at gmail.com Thu Jun 8 07:15:38 2017 From: s.safarov at gmail.com (Sergey Safarov) Date: Thu, 08 Jun 2017 07:15:38 +0000 Subject: [Freeswitch-users] BugHunt Message-ID: Is BugHunt will be today? -------------- next part -------------- An HTML attachment was scrubbed... URL: From denys.pozniak at crazycall.com Thu Jun 8 08:07:18 2017 From: denys.pozniak at crazycall.com (Denys Pozniak) Date: Thu, 8 Jun 2017 10:07:18 +0200 Subject: [Freeswitch-users] question about HA solution In-Reply-To: References: <8A13E0EC-FA13-4267-80F6-CE1A8E8360CF@jerris.com> Message-ID: Hello! My configs: *switch.conf.xml* *external.conf.xml* On 7 June 2017 at 17:35, Michael Jerris wrote: > That param should keep it from doing so, if its not you are not setting it > somehow or something else is wiping the db. > > On Jun 5, 2017, at 1:50 PM, Denys Pozniak > wrote: > > Yes, correct. But when you restart FS on slave, it will erase database. > And option auto-clear-sql=false not working for me. > > On Jun 5, 2017 6:32 PM, "Michael Jerris" wrote: > > recovered calls will get new entries in the table. > > On Jun 5, 2017, at 7:41 AM, Denys Pozniak > wrote: > > Hello! > > Thank you *Raymond* about your explanation, but I dont agree with some > point: > *If it really need an answer about your question -- "if it is possible to > move calls back". I think it's unnecessary.* - in my case I have two > not equal servers, so I need to have only one as a master. > If switchover happens I need to have ability to restore master back. > > Thank you *Luis* for your link, you can do simple test to understand what > I am talking about: do call -> check on master and slave #show channels -> > restart FS on slave -> check on master #show channels. In my case I dont > see any active calls after this, so restoring back is not possible. > > > > On 3 June 2017 at 22:16, Luis Daniel Lucio Quiroz < > luis.daniel.lucio at gmail.com> wrote: > >> You may want to read this article. >> >> http://inside-out.xyz/technology/how-to-configure-freeswitch-for-ha.html >> >> Le 31 mai 2017 6:29 PM, "Denys Pozniak" a >> écrit : >> >> Hello! >> >> I built FS HA solution based on keepalived and mysql master-master. >> It works ok generally, but as I understand FS after restarting cleaning >> own database. >> >> So when node1 fails calls jump to node2, after script restarts node1 it >> is not possible to move calls back. >> >> Tried options in switch.conf.xml, but no luck: >> >> >> >> >> Is there is a way to solve this? >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From v.zakhozhai at gmail.com Thu Jun 8 09:07:26 2017 From: v.zakhozhai at gmail.com (Vladyslav Zakhozhai) Date: Thu, 8 Jun 2017 12:07:26 +0300 Subject: [Freeswitch-users] XML curl "not found" response Message-ID: Hi, I am curious what HTTP status xml curl server must provide when no result is found for request with body
I mean HTTP 200 or HTTP 404. My problem is the following: 1. When freeswitch gets 404 for directory request it reports error: [ERR] mod_xml_curl.c:315 Received HTTP error 404 trying to fetch 2. When freeswitch gets 200 with status "not found" on dialplan request failover to file xml config does not occure. Thank you in advance. -- С уважением, Владислав Захожай -------------- next part -------------- An HTML attachment was scrubbed... URL: From gmaruzz at gmail.com Thu Jun 8 09:18:43 2017 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Thu, 8 Jun 2017 11:18:43 +0200 Subject: [Freeswitch-users] XML curl "not found" response In-Reply-To: References: Message-ID: """If it receives a valid response from your web application, then it will load the configuration just like it would if you had put it into the FreeSWITCH Configuration File . If it receives an invalid or 404 *not found* response, then it will attempt to look for the file on disk instead.""" """https://freeswitch.org/confluence/display/FREESWITCH/ mod_xml_curl#mod_xml_curl-Section:notfound""" https://freeswitch.org/confluence/display/FREESWITCH/mod_xml_curl On 8 June 2017 at 11:07, Vladyslav Zakhozhai wrote: > Hi, > > I am curious what HTTP status xml curl server must provide when no result > is found for request with body > > > >
> >
> > > > I mean HTTP 200 or HTTP 404. > > > My problem is the following: > > 1. When freeswitch gets 404 for directory request it reports error: > > [ERR] mod_xml_curl.c:315 Received HTTP error 404 trying to fetch > > > 2. When freeswitch gets 200 with status "not found" on dialplan request > failover to file xml config does not occure. > > Thank you in advance. > > -- > С уважением, > Владислав Захожай > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at 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 OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: From mase2hot at gmail.com Thu Jun 8 09:40:48 2017 From: mase2hot at gmail.com (Jason Bedward) Date: Thu, 8 Jun 2017 10:40:48 +0100 Subject: [Freeswitch-users] Re-invite Glare issue - FS ending call In-Reply-To: References: <72A53E3A-47FC-498E-AF90-2B39448DE52A@jerris.com> Message-ID: Looking again as this according to RFC 3261 Freeswitch is doing this wrong? For the B leg my FS is the UAC, and if so when the invite comes in from the B leg it shouldn't respond 491 as that should only be done from the UAS. Unless I'm wrong with this? On Thu, Jun 8, 2017 at 8:14 AM, Jason Bedward wrote: > My provider is sending the reinvite on the A leg FS the sends this on to > the B leg. At same time the B leg provider sends a reinvite. > > I dont believe FS is sending the reinvite on its own accord. Its only > because of the A leg and that we are using bypass media. So I dont think > session timers will make a difference. > > FYI A leg and B leg provider are the same company although different > servers. > > On 7 Jun 2017 16:57, "Michael Jerris" wrote: > >> why are they both sending a re-invite at the same time? This is called >> glare, we seem to be handling it properly, the provider seems not to be. >> The easiest fix here is probably to figure out why we are both sending >> re-invite at the same time, maybe session timers and the provider is broken >> and trying to send it when we said we would. >> >> On Jun 7, 2017, at 11:49 AM, Jason Bedward wrote: >> >> Hi, >> >> I have an issue with re-invites on some calls. I'm using 1.6.15 and these >> calls are using bypass media. I have Kamailio as inbound SBC but FS >> connecting directly to my provider for B leg. >> >> >> - After 5 minutes plus on some calls my provider sends re-invite on A >> leg >> - FS then sends this re-invite to the B leg >> - At the same time the B leg sends a Re-invite >> - FS replys 491 >> - B leg provider replys 100, then 500 (with retry in the 500) >> - FS send ACK and then BYE >> >> Not sure what setting I need to change or can change infact to either >> retry the invite in accordance with the 500 retry request. Or something >> else to stop the call ending... >> >> >> >> Thanks >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at 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 at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: From v.zakhozhai at gmail.com Thu Jun 8 11:43:47 2017 From: v.zakhozhai at gmail.com (Vladyslav Zakhozhai) Date: Thu, 8 Jun 2017 14:43:47 +0300 Subject: [Freeswitch-users] XML curl "not found" response In-Reply-To: References: Message-ID: Giovanni, thank you for your reply. Yes of course, I've read it. And the main question is - what is appropriate way to inform freeswitch with "not found". And once more why freeswitch treats 404 response as an error? E.g. freeswitch requests configuration for non-existent user. Here we have two options: 1. Freeswitch replies with XML not found and HTTP 200. Freeswitch understands it pretty correct and in logs I can see that freeswitch warns me it was not able to find user in some domain. This is ok and correct. 2. Freeswitch replies with XML not found and HTTP 404. Freeswitch reports an error. It do not wait HTTP 404 status code. And once more about p.2 from freeswitch's log: "[ERR] mod_xml_curl.c:315 Received HTTP error 404 trying to fetch ....." 2017-06-08 12:18 GMT+03:00 Giovanni Maruzzelli : > """If it receives a valid response from your web application, then it will > load the configuration just like it would if you had put it into the FreeSWITCH > Configuration File . If > it receives an invalid or 404 *not found* response, then it will attempt > to look for the file on disk instead.""" > > """https://freeswitch.org/confluence/display/FREESWITCH/mod_ > xml_curl#mod_xml_curl-Section:notfound""" > > https://freeswitch.org/confluence/display/FREESWITCH/mod_xml_curl > > > On 8 June 2017 at 11:07, Vladyslav Zakhozhai > wrote: > >> Hi, >> >> I am curious what HTTP status xml curl server must provide when no result >> is found for request with body >> >> >> >>
>> >>
>> >> >> >> I mean HTTP 200 or HTTP 404. >> >> >> My problem is the following: >> >> 1. When freeswitch gets 404 for directory request it reports error: >> >> [ERR] mod_xml_curl.c:315 Received HTTP error 404 trying to fetch >> >> >> 2. When freeswitch gets 200 with status "not found" on dialplan request >> failover to file xml config does not occure. >> >> Thank you in advance. >> >> -- >> С уважением, >> Владислав Захожай >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at 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 > OpenTelecom.IT > cell: +39 347 266 56 18 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- С уважением, Владислав Захожай -------------- next part -------------- An HTML attachment was scrubbed... URL: From vma at vallimamod.org Thu Jun 8 12:53:15 2017 From: vma at vallimamod.org (Vallimamod Abdullah) Date: Thu, 8 Jun 2017 14:53:15 +0200 Subject: [Freeswitch-users] XML curl "not found" response In-Reply-To: References: Message-ID: Hi, The correct reply is HTTP 200. Any non 200 body is discarded by mod_xml_curl and triggers a failover to a backup url if you have defined one or the static conf. If the failover to static conf does not occur in your case with a 200 'not found' reply, you need to investigate why. Can you post any full debug log? Best Regards, -- Vallimamod Abdullah SIP Solutions VOIP Consulting vma at sipsolutions.fr . > On 8 Jun 2017, at 13:43, Vladyslav Zakhozhai wrote: > > Giovanni, thank you for your reply. > > Yes of course, I've read it. And the main question is - what is appropriate way to inform freeswitch with "not found". And once more why freeswitch treats 404 response as an error? > > E.g. freeswitch requests configuration for non-existent user. Here we have two options: > 1. Freeswitch replies with XML not found and HTTP 200. Freeswitch understands it pretty correct and in logs I can see that freeswitch warns me it was not able to find user in some domain. This is ok and correct. > > 2. Freeswitch replies with XML not found and HTTP 404. Freeswitch reports an error. It do not wait HTTP 404 status code. > > And once more about p.2 from freeswitch's log: > > "[ERR] mod_xml_curl.c:315 Received HTTP error 404 trying to fetch ....." > > > 2017-06-08 12:18 GMT+03:00 Giovanni Maruzzelli : > """If it receives a valid response from your web application, then it will load the configuration just like it would if you had put it into the FreeSWITCH Configuration File. If it receives an invalid or 404 not found response, then it will attempt to look for the file on disk instead.""" > > """https://freeswitch.org/confluence/display/FREESWITCH/mod_xml_curl#mod_xml_curl-Section:notfound""" > > https://freeswitch.org/confluence/display/FREESWITCH/mod_xml_curl > > > On 8 June 2017 at 11:07, Vladyslav Zakhozhai wrote: > Hi, > > I am curious what HTTP status xml curl server must provide when no result is found for request with body > > > >
> >
> > > > I mean HTTP 200 or HTTP 404. > > > My problem is the following: > > 1. When freeswitch gets 404 for directory request it reports error: > > [ERR] mod_xml_curl.c:315 Received HTTP error 404 trying to fetch > > > 2. When freeswitch gets 200 with status "not found" on dialplan request failover to file xml config does not occure. > > Thank you in advance. > > -- > С уважением, > Владислав Захожай From v.zakhozhai at gmail.com Thu Jun 8 14:30:39 2017 From: v.zakhozhai at gmail.com (Vladyslav Zakhozhai) Date: Thu, 8 Jun 2017 17:30:39 +0300 Subject: [Freeswitch-users] XML curl "not found" response In-Reply-To: References: Message-ID: Vallimamod, Thank you for you clear answer. I appreciate that. I though that 200 "not found" is correct but I had some doubts. You cleared out this question for me. I'll play with dialplan and failover to XML static config little bit later and give you feedback. 2017-06-08 15:53 GMT+03:00 Vallimamod Abdullah : > Hi, > > The correct reply is HTTP 200. > Any non 200 body is discarded by mod_xml_curl and triggers a failover to a > backup url if you have defined one or the static conf. > > If the failover to static conf does not occur in your case with a 200 'not > found' reply, you need to investigate why. Can you post any full debug log? > > Best Regards, > -- > Vallimamod Abdullah > SIP Solutions > VOIP Consulting > vma at sipsolutions.fr > . > > > On 8 Jun 2017, at 13:43, Vladyslav Zakhozhai > wrote: > > > > Giovanni, thank you for your reply. > > > > Yes of course, I've read it. And the main question is - what is > appropriate way to inform freeswitch with "not found". And once more why > freeswitch treats 404 response as an error? > > > > E.g. freeswitch requests configuration for non-existent user. Here we > have two options: > > 1. Freeswitch replies with XML not found and HTTP 200. Freeswitch > understands it pretty correct and in logs I can see that freeswitch warns > me it was not able to find user in some domain. This is ok and correct. > > > > 2. Freeswitch replies with XML not found and HTTP 404. Freeswitch > reports an error. It do not wait HTTP 404 status code. > > > > And once more about p.2 from freeswitch's log: > > > > "[ERR] mod_xml_curl.c:315 Received HTTP error 404 trying to fetch ....." > > > > > > 2017-06-08 12:18 GMT+03:00 Giovanni Maruzzelli : > > """If it receives a valid response from your web application, then it > will load the configuration just like it would if you had put it into the > FreeSWITCH Configuration File. If it receives an invalid or 404 not found > response, then it will attempt to look for the file on disk instead.""" > > > > """https://freeswitch.org/confluence/display/FREESWITCH/ > mod_xml_curl#mod_xml_curl-Section:notfound""" > > > > https://freeswitch.org/confluence/display/FREESWITCH/mod_xml_curl > > > > > > On 8 June 2017 at 11:07, Vladyslav Zakhozhai > wrote: > > Hi, > > > > I am curious what HTTP status xml curl server must provide when no > result is found for request with body > > > > > > > >
> > > >
> > > > > > > > I mean HTTP 200 or HTTP 404. > > > > > > My problem is the following: > > > > 1. When freeswitch gets 404 for directory request it reports error: > > > > [ERR] mod_xml_curl.c:315 Received HTTP error 404 trying to fetch > > > > > > 2. When freeswitch gets 200 with status "not found" on dialplan request > failover to file xml config does not occure. > > > > Thank you in advance. > > > > -- > > С уважением, > > Владислав Захожай > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- С уважением, Владислав Захожай -------------- next part -------------- An HTML attachment was scrubbed... URL: From richardm at cellularity.co.uk Thu Jun 8 10:33:44 2017 From: richardm at cellularity.co.uk (Richard Melville) Date: Thu, 8 Jun 2017 11:33:44 +0100 Subject: [Freeswitch-users] Freeswitch Dependencies Message-ID: I haven't used or built Freeswitch yet, but I'm getting closer to that point. I will be building from source. Having looked at the documentation I can see that Debian is preferred, and the use of distros in general. However, I don't use distros, but rather build my own systems from scratch. I've looked at the Centos page ( https://freeswitch.org/ confluence/display/FREESWITCH/CentOS+7+and+RHEL+7 ) and under "Building from source" there appears to be a list of dependencies. I already have most of those dependencies installed (other than the codecs) but I have three questions which maybe somebody can answer. The first is: can openssl be substituted with libressl, which I use? The second is: "mongo-c-driver-devel" suggests that mongodb is a dependency of Freeswitch. I've seen no mention of mongodb anywhere in either the book, or the documentation generally, so why is this listed as a dependency? The third is: there is no mention of package version numbers anywhere, so how can I find if there are any issues with particular versions? Any help gratefully received. -- Richard Melville Systems Architect cellularity.co.uk stellarsystem.wordpress.com +44 20 33 555 305 +44 7957 836330 -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Thu Jun 8 15:15:47 2017 From: mike at jerris.com (Michael Jerris) Date: Thu, 08 Jun 2017 15:15:47 +0000 Subject: [Freeswitch-users] BugHunt In-Reply-To: References: Message-ID: if you have things to discuss we can... ping me on hipchat and i'll call in On Thu, Jun 8, 2017 at 3:16 AM Sergey Safarov wrote: > Is BugHunt will be today? > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Thu Jun 8 15:20:29 2017 From: mike at jerris.com (Michael Jerris) Date: Thu, 08 Jun 2017 15:20:29 +0000 Subject: [Freeswitch-users] Re-invite Glare issue - FS ending call In-Reply-To: References: <72A53E3A-47FC-498E-AF90-2B39448DE52A@jerris.com> Message-ID: the 491 is in response to their re-invite, in which we are the UAS. For our re-invite we send we are UAC. On Thu, Jun 8, 2017 at 5:41 AM Jason Bedward wrote: > Looking again as this according to RFC 3261 Freeswitch is doing this wrong? > > For the B leg my FS is the UAC, and if so when the invite comes in from > the B leg it shouldn't respond 491 as that should only be done from the > UAS. > > Unless I'm wrong with this? > > On Thu, Jun 8, 2017 at 8:14 AM, Jason Bedward wrote: > >> My provider is sending the reinvite on the A leg FS the sends this on to >> the B leg. At same time the B leg provider sends a reinvite. >> >> I dont believe FS is sending the reinvite on its own accord. Its only >> because of the A leg and that we are using bypass media. So I dont think >> session timers will make a difference. >> >> FYI A leg and B leg provider are the same company although different >> servers. >> >> On 7 Jun 2017 16:57, "Michael Jerris" wrote: >> >>> why are they both sending a re-invite at the same time? This is called >>> glare, we seem to be handling it properly, the provider seems not to be. >>> The easiest fix here is probably to figure out why we are both sending >>> re-invite at the same time, maybe session timers and the provider is broken >>> and trying to send it when we said we would. >>> >>> On Jun 7, 2017, at 11:49 AM, Jason Bedward wrote: >>> >>> Hi, >>> >>> I have an issue with re-invites on some calls. I'm using 1.6.15 and >>> these calls are using bypass media. I have Kamailio as inbound SBC but FS >>> connecting directly to my provider for B leg. >>> >>> >>> - After 5 minutes plus on some calls my provider sends re-invite on >>> A leg >>> - FS then sends this re-invite to the B leg >>> - At the same time the B leg sends a Re-invite >>> - FS replys 491 >>> - B leg provider replys 100, then 500 (with retry in the 500) >>> - FS send ACK and then BYE >>> >>> Not sure what setting I need to change or can change infact to either >>> retry the invite in accordance with the 500 retry request. Or something >>> else to stop the call ending... >>> >>> >>> >>> Thanks >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at 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 at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From bipin at xbipin.com Thu Jun 8 15:41:52 2017 From: bipin at xbipin.com (Bipin Patel) Date: Thu, 8 Jun 2017 19:41:52 +0400 Subject: [Freeswitch-users] weird dtmf forwarding issue Message-ID: hi, i have 2 setups running FS with the exact same profile and dialplan, one running on windows and the other on a raspberry pi, both receive calls from registered clients and forward to a remote gateway, the problem is when client sends dtmf, the windows FS forwards to gateway without issues but on the rpi most of the times fs isnt able to detect dtmf or at times detects but doesnt or partially sends to gateway. i have been banging my head with this from a few days but no idea whats going wrong, the client sending to rpi FS i even made him send call to gateway directly and then that works so this rules out client issue, no idea whats wrong with FS, both are on the latest build im using g711u throughout -- Regards, Bipin ------------------------------------------------------------------------ -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Thu Jun 8 15:48:57 2017 From: mike at jerris.com (Michael Jerris) Date: Thu, 08 Jun 2017 15:48:57 +0000 Subject: [Freeswitch-users] question about HA solution In-Reply-To: References: <8A13E0EC-FA13-4267-80F6-CE1A8E8360CF@jerris.com> Message-ID: check your db logs as nothing we are doing should be clearing those. On Thu, Jun 8, 2017 at 4:08 AM Denys Pozniak wrote: > Hello! > > My configs: > > *switch.conf.xml* > > > value="odbc://freeswitch:root:ubuntu"/> > > > > > > *external.conf.xml* > > > > > > On 7 June 2017 at 17:35, Michael Jerris wrote: > >> That param should keep it from doing so, if its not you are not setting >> it somehow or something else is wiping the db. >> >> On Jun 5, 2017, at 1:50 PM, Denys Pozniak >> wrote: >> >> Yes, correct. But when you restart FS on slave, it will erase database. >> And option auto-clear-sql=false not working for me. >> >> On Jun 5, 2017 6:32 PM, "Michael Jerris" wrote: >> >> recovered calls will get new entries in the table. >> >> On Jun 5, 2017, at 7:41 AM, Denys Pozniak >> wrote: >> >> Hello! >> >> Thank you *Raymond* about your explanation, but I dont agree with some >> point: >> *If it really need an answer about your question -- "if it is possible to >> move calls back". I think it's unnecessary.* - in my case I have two >> not equal servers, so I need to have only one as a master. >> If switchover happens I need to have ability to restore master back. >> >> Thank you *Luis* for your link, you can do simple test to understand >> what I am talking about: do call -> check on master and slave #show >> channels -> restart FS on slave -> check on master #show channels. In my >> case I dont see any active calls after this, so restoring back is not >> possible. >> >> >> >> On 3 June 2017 at 22:16, Luis Daniel Lucio Quiroz < >> luis.daniel.lucio at gmail.com> wrote: >> >>> You may want to read this article. >>> >>> http://inside-out.xyz/technology/how-to-configure-freeswitch-for-ha.html >>> >>> Le 31 mai 2017 6:29 PM, "Denys Pozniak" a >>> écrit : >>> >>> Hello! >>> >>> I built FS HA solution based on keepalived and mysql master-master. >>> It works ok generally, but as I understand FS after restarting cleaning >>> own database. >>> >>> So when node1 fails calls jump to node2, after script restarts node1 it >>> is not possible to move calls back. >>> >>> Tried options in switch.conf.xml, but no luck: >>> >>> >>> >>> >>> Is there is a way to solve this? >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Thu Jun 8 16:01:54 2017 From: mike at jerris.com (Michael Jerris) Date: Thu, 8 Jun 2017 12:01:54 -0400 Subject: [Freeswitch-users] Freeswitch Dependencies In-Reply-To: References: Message-ID: <091C458B-44F6-42F9-90A8-FFACEFC534D5@jerris.com> > On Jun 8, 2017, at 6:33 AM, Richard Melville wrote: > > I haven't used or built Freeswitch yet, but I'm getting closer to that point. I will be building from source. Having looked at the documentation I can see that Debian is preferred, and the use of distros in general. However, I don't use distros, but rather build my own systems from scratch. > > I've looked at the Centos page ( https://freeswitch.org/confluence/display/FREESWITCH/CentOS+7+and+RHEL+7 ) and under "Building from source" there appears to be a list of dependencies. I already have most of those dependencies installed (other than the codecs) but I have three questions which maybe somebody can answer. > > The first is: can openssl be substituted with libressl, which I use? Not sure, depends if libressl has the required pieces we need for dtls-srtp and all the required ciphers required by the browsers for webrtc. > > The second is: "mongo-c-driver-devel" suggests that mongodb is a dependency of Freeswitch. I've seen no mention of mongodb anywhere in either the book, or the documentation generally, so why is this listed as a dependency? Its a module, if you don’t want that module, its not needed. > > The third is: there is no mention of package version numbers anywhere, so how can I find if there are any issues with particular versions? We don’t test a vast array of different package versions, we do testing based on the ones for the distros we package for. The versions in Debian 8 are well tested, other versions are much less well tested or not tested at all. As for other libs, use the ones in our stash project for dep libs when not otherwise available. Creating extensive documentation for building on your own distro would be far more work than even adding support for a new distro, and we don’t have any plans to create that. I’m happy to respond to some specific questions, but there are limits to the amount of time that it makes sense for us to spend on issues like this for a single person. > > Any help gratefully received. > -------------- next part -------------- An HTML attachment was scrubbed... URL: From v.zakhozhai at gmail.com Thu Jun 8 16:05:31 2017 From: v.zakhozhai at gmail.com (Vladyslav Zakhozhai) Date: Thu, 8 Jun 2017 19:05:31 +0300 Subject: [Freeswitch-users] XML curl "not found" response In-Reply-To: References: Message-ID: That was my mistake. With HTTP 404 dialplan fails over to local static XML Dialplan. Thank you all. 2017-06-08 17:30 GMT+03:00 Vladyslav Zakhozhai : > Vallimamod, > > Thank you for you clear answer. I appreciate that. I though that 200 "not > found" is correct but I had some doubts. You cleared out this question for > me. > > I'll play with dialplan and failover to XML static config little bit later > and give you feedback. > > > 2017-06-08 15:53 GMT+03:00 Vallimamod Abdullah : > >> Hi, >> >> The correct reply is HTTP 200. >> Any non 200 body is discarded by mod_xml_curl and triggers a failover to >> a backup url if you have defined one or the static conf. >> >> If the failover to static conf does not occur in your case with a 200 >> 'not found' reply, you need to investigate why. Can you post any full debug >> log? >> >> Best Regards, >> -- >> Vallimamod Abdullah >> SIP Solutions >> VOIP Consulting >> vma at sipsolutions.fr >> . >> >> > On 8 Jun 2017, at 13:43, Vladyslav Zakhozhai >> wrote: >> > >> > Giovanni, thank you for your reply. >> > >> > Yes of course, I've read it. And the main question is - what is >> appropriate way to inform freeswitch with "not found". And once more why >> freeswitch treats 404 response as an error? >> > >> > E.g. freeswitch requests configuration for non-existent user. Here we >> have two options: >> > 1. Freeswitch replies with XML not found and HTTP 200. Freeswitch >> understands it pretty correct and in logs I can see that freeswitch warns >> me it was not able to find user in some domain. This is ok and correct. >> > >> > 2. Freeswitch replies with XML not found and HTTP 404. Freeswitch >> reports an error. It do not wait HTTP 404 status code. >> > >> > And once more about p.2 from freeswitch's log: >> > >> > "[ERR] mod_xml_curl.c:315 Received HTTP error 404 trying to fetch ....." >> > >> > >> > 2017-06-08 12:18 GMT+03:00 Giovanni Maruzzelli : >> > """If it receives a valid response from your web application, then it >> will load the configuration just like it would if you had put it into the >> FreeSWITCH Configuration File. If it receives an invalid or 404 not found >> response, then it will attempt to look for the file on disk instead.""" >> > >> > """https://freeswitch.org/confluence/display/FREESWITCH/mod_ >> xml_curl#mod_xml_curl-Section:notfound""" >> > >> > https://freeswitch.org/confluence/display/FREESWITCH/mod_xml_curl >> > >> > >> > On 8 June 2017 at 11:07, Vladyslav Zakhozhai >> wrote: >> > Hi, >> > >> > I am curious what HTTP status xml curl server must provide when no >> result is found for request with body >> > >> > >> > >> >
>> > >> >
>> > >> > >> > >> > I mean HTTP 200 or HTTP 404. >> > >> > >> > My problem is the following: >> > >> > 1. When freeswitch gets 404 for directory request it reports error: >> > >> > [ERR] mod_xml_curl.c:315 Received HTTP error 404 trying to fetch >> > >> > >> > 2. When freeswitch gets 200 with status "not found" on dialplan request >> failover to file xml config does not occure. >> > >> > Thank you in advance. >> > >> > -- >> > С уважением, >> > Владислав Захожай >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > С уважением, > Владислав Захожай > > -- С уважением, Владислав Захожай -------------- next part -------------- An HTML attachment was scrubbed... URL: From anthony.minessale at gmail.com Thu Jun 8 16:16:12 2017 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 08 Jun 2017 16:16:12 +0000 Subject: [Freeswitch-users] Re-invite Glare issue - FS ending call In-Reply-To: References: <72A53E3A-47FC-498E-AF90-2B39448DE52A@jerris.com> Message-ID: As always dont report issues via email. https://freeswitch.org/jira On Thu, Jun 8, 2017 at 10:21 AM Michael Jerris wrote: > the 491 is in response to their re-invite, in which we are the UAS. For > our re-invite we send we are UAC. > > On Thu, Jun 8, 2017 at 5:41 AM Jason Bedward wrote: > >> Looking again as this according to RFC 3261 Freeswitch is doing this >> wrong? >> >> For the B leg my FS is the UAC, and if so when the invite comes in from >> the B leg it shouldn't respond 491 as that should only be done from the >> UAS. >> >> Unless I'm wrong with this? >> >> On Thu, Jun 8, 2017 at 8:14 AM, Jason Bedward wrote: >> >>> My provider is sending the reinvite on the A leg FS the sends this on to >>> the B leg. At same time the B leg provider sends a reinvite. >>> >>> I dont believe FS is sending the reinvite on its own accord. Its only >>> because of the A leg and that we are using bypass media. So I dont think >>> session timers will make a difference. >>> >>> FYI A leg and B leg provider are the same company although different >>> servers. >>> >>> On 7 Jun 2017 16:57, "Michael Jerris" wrote: >>> >>>> why are they both sending a re-invite at the same time? This is called >>>> glare, we seem to be handling it properly, the provider seems not to be. >>>> The easiest fix here is probably to figure out why we are both sending >>>> re-invite at the same time, maybe session timers and the provider is broken >>>> and trying to send it when we said we would. >>>> >>>> On Jun 7, 2017, at 11:49 AM, Jason Bedward wrote: >>>> >>>> Hi, >>>> >>>> I have an issue with re-invites on some calls. I'm using 1.6.15 and >>>> these calls are using bypass media. I have Kamailio as inbound SBC but FS >>>> connecting directly to my provider for B leg. >>>> >>>> >>>> - After 5 minutes plus on some calls my provider sends re-invite on >>>> A leg >>>> - FS then sends this re-invite to the B leg >>>> - At the same time the B leg sends a Re-invite >>>> - FS replys 491 >>>> - B leg provider replys 100, then 500 (with retry in the 500) >>>> - FS send ACK and then BYE >>>> >>>> Not sure what setting I need to change or can change infact to either >>>> retry the invite in accordance with the 500 retry request. Or something >>>> else to stop the call ending... >>>> >>>> >>>> >>>> Thanks >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 ♬ @anthmfs ♬ @FreeSWITCH ♬ ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ * ClueCon Weekly Development Call ☎ sip:888 at conference.freeswitch.org ☎ +19193869900 https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA -------------- next part -------------- An HTML attachment was scrubbed... URL: From krice at freeswitch.org Thu Jun 8 17:03:39 2017 From: krice at freeswitch.org (Ken Rice) Date: Thu, 8 Jun 2017 12:03:39 -0500 Subject: [Freeswitch-users] Freeswitch Dependencies In-Reply-To: References: Message-ID: <49cb01d2e079$2dc6beb0$89543c10$@freeswitch.org> To partially address some of your questions. 1. The reason we recommend specific distros as it ends up being a known quantity from a supportability stand point. This means that we test builds on those distros with what they provide as versions for system supplied packages. We highly recommend Debian8 for the least pain as that is where primary development and bug fixes occur and from there they are ported to address any differences in what other systems support. Use of distros in general are encouraged for this reason. The one off support for a custom rolled system is beyond the capabilities of the project to effectively support with our small team. 2. As far as libressl vs openssl. I don’t know of anyone actually testing this to see if it will work or not. We specifically test against versions of OpenSSL. The later versions of OpenSSL are required due not only to the security concerns you are trying to address with LibreSSL, but due to newer encryption technologies being included in the later versions of OpenSSL that may or may not be included in LibreSSL. (again this comes down to supporting what is available and most widely used on the supported distributions.) 3. As far as mongo-c-driver-devel, you will find that there are several dependancies like that may or may not be required for you to build FreeSWITCH depending on which modules available that you may want to build. While some of the deps are required to build the core (such as sqlite et al), however others are only required if you plan on building a module that requires them such as mod_mongo. Other examples are mod_flite, mod_cepstral, mod_hiredis, mod_osp and mod_ladspa. (mod_ladspa itself is an example of a module that only works on Linux unless someone gotten around to porting this linux specific interface over to Windows and possible some of the BSDs) Now that being said you can review the From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Richard Melville Sent: Thursday, June 8, 2017 5:34 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Freeswitch Dependencies I haven't used or built Freeswitch yet, but I'm getting closer to that point. I will be building from source. Having looked at the documentation I can see that Debian is preferred, and the use of distros in general. However, I don't use distros, but rather build my own systems from scratch. I've looked at the Centos page ( https://freeswitch.org/confluence/display/FREESWITCH/CentOS+7+and+RHEL+7 ) and under "Building from source" there appears to be a list of dependencies. I already have most of those dependencies installed (other than the codecs) but I have three questions which maybe somebody can answer. The first is: can openssl be substituted with libressl, which I use? The second is: "mongo-c-driver-devel" suggests that mongodb is a dependency of Freeswitch. I've seen no mention of mongodb anywhere in either the book, or the documentation generally, so why is this listed as a dependency? The third is: there is no mention of package version numbers anywhere, so how can I find if there are any issues with particular versions? Any help gratefully received. -- Richard Melville Systems Architect cellularity.co.uk stellarsystem.wordpress.com +44 20 33 555 305 +44 7957 836330 -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian at freeswitch.org Thu Jun 8 20:23:58 2017 From: brian at freeswitch.org (Brian West) Date: Thu, 8 Jun 2017 15:23:58 -0500 Subject: [Freeswitch-users] weird dtmf forwarding issue In-Reply-To: References: Message-ID: Its a little unclear, you say 'detect', so I have to assume you're running a dtmf detector. You're running on x86 vs arm, so I can only assume its possibly that differences in architecture are at play here. /b On Thu, Jun 8, 2017 at 10:41 AM, Bipin Patel wrote: > hi, > > i have 2 setups running FS with the exact same profile and dialplan, one > running on windows and the other on a raspberry pi, both receive calls from > registered clients and forward to a remote gateway, the problem is when > client sends dtmf, the windows FS forwards to gateway without issues but on > the rpi most of the times fs isnt able to detect dtmf or at times detects > but doesnt or partially sends to gateway. > i have been banging my head with this from a few days but no idea whats > going wrong, the client sending to rpi FS i even made him send call to > gateway directly and then that works so this rules out client issue, no > idea whats wrong with FS, both are on the latest build > > im using g711u throughout > > -- > Regards, > Bipin > > > ------------------------------ > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Book a phone call (CST) Allison prompts for FreeSWITCH: *https://www.gofundme.com/allison-prompts-for-freeswitch* Got Bugs? Report them here ! | Reddit: /r/freeswitch *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: From bipin at xbipin.com Thu Jun 8 21:10:31 2017 From: bipin at xbipin.com (Bipin Patel) Date: Fri, 09 Jun 2017 01:10:31 +0400 Subject: [Freeswitch-users] weird dtmf forwarding issue In-Reply-To: References: Message-ID: <15c898b5d58.2779.b07ebdf329620b8089087c7205b03f01@xbipin.com> Detect as in when fs receives a DTMF it prints in cli, when it doesn't it didn't detect a DTMF even though client is sending it. First I tried sending inband and used start DTMF but that wasn't too good so moved to the rfc one and then finally ended up using info. I made some progress on this, wanted to know can fs receive DTMF as rfc or info and send info or rfc to gateway, something like a intercept and forward as a new method? When I set DTMF to rfc in profile and client sends in info then nothing goes to gateway, if client sends info and I set info in profile then it works. In summary I want to accept info and rfc and want to send to gateway in whatever mode they support rather than force client to send in a particular format. Also is there a way to make DTMF timings a little relaxed, I have another client that uses fs as PBX and calls come to it using a pstn to VoIP gateway and due to the exchange under going hardware upgrades the DTMF timings are a little out or the trailing sound is a bit broken and most of the times the pstn gateway doesn't detect so I switched it to inband and if I can tune fs to accept a little messed up DTMF then it would make it work. I had a similar issue in a different country when the exchange was moving to etsi fsk for caller ID and it took months for the company to figure out the issue and sort it out On June 9, 2017 12:27:04 AM Brian West wrote: > Its a little unclear, you say 'detect', so I have to assume you're running > a dtmf detector. You're running on x86 vs arm, so I can only assume its > possibly that differences in architecture are at play here. > > /b > > > On Thu, Jun 8, 2017 at 10:41 AM, Bipin Patel wrote: > >> hi, >> >> i have 2 setups running FS with the exact same profile and dialplan, one >> running on windows and the other on a raspberry pi, both receive calls from >> registered clients and forward to a remote gateway, the problem is when >> client sends dtmf, the windows FS forwards to gateway without issues but on >> the rpi most of the times fs isnt able to detect dtmf or at times detects >> but doesnt or partially sends to gateway. >> i have been banging my head with this from a few days but no idea whats >> going wrong, the client sending to rpi FS i even made him send call to >> gateway directly and then that works so this rules out client issue, no >> idea whats wrong with FS, both are on the latest build >> >> im using g711u throughout >> >> -- >> Regards, >> Bipin >> >> >> ------------------------------ >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > > *Brian West* > brian at freeswitch.org > > *Twitter: @FreeSWITCH , @briankwest* > > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > Book a phone call (CST) > > Allison prompts for FreeSWITCH: > > *https://www.gofundme.com/allison-prompts-for-freeswitch* > > > Got Bugs? Report them here ! | Reddit: > /r/freeswitch > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *Skype:*briankwest > > > > ---------- > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From krice at freeswitch.org Thu Jun 8 21:34:07 2017 From: krice at freeswitch.org (Ken Rice) Date: Thu, 8 Jun 2017 16:34:07 -0500 Subject: [Freeswitch-users] Freeswitch Dependencies In-Reply-To: References: Message-ID: <4bdb01d2e09e$f60fa240$e22ee6c0$@freeswitch.org> And as usual I got interrupted and clicked send before finishing a thought… (sorry about that but it helps demonstrate our time constraints) But following up my earlier comments Now that being said you can review the freeswitch.spec file in the root of the source tree or the Debian packaging files in the debian dir also in the source tree to get a list of build deps and if min versions are required what those are. Things without version numbering in those files are that way due to the current versions in the appropriate Distros being sufficient and Distros typically do not change APIs once theyhave released a version (or on minor versions upgrades)_ From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Richard Melville Sent: Thursday, June 8, 2017 5:34 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Freeswitch Dependencies I haven't used or built Freeswitch yet, but I'm getting closer to that point. I will be building from source. Having looked at the documentation I can see that Debian is preferred, and the use of distros in general. However, I don't use distros, but rather build my own systems from scratch. I've looked at the Centos page ( https://freeswitch.org/confluence/display/FREESWITCH/CentOS+7+and+RHEL+7 ) and under "Building from source" there appears to be a list of dependencies. I already have most of those dependencies installed (other than the codecs) but I have three questions which maybe somebody can answer. The first is: can openssl be substituted with libressl, which I use? The second is: "mongo-c-driver-devel" suggests that mongodb is a dependency of Freeswitch. I've seen no mention of mongodb anywhere in either the book, or the documentation generally, so why is this listed as a dependency? The third is: there is no mention of package version numbers anywhere, so how can I find if there are any issues with particular versions? Any help gratefully received. -- Richard Melville Systems Architect cellularity.co.uk stellarsystem.wordpress.com +44 20 33 555 305 +44 7957 836330 -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Fri Jun 9 05:48:21 2017 From: david.villasmil.work at gmail.com (David Villasmil) Date: Fri, 9 Jun 2017 01:48:21 -0400 Subject: [Freeswitch-users] Removing userpart of contact Message-ID: Hello guys, I ran into a situation where I need the contact to be like: Meaning I need to remove the username, i've trying doing this but FS adds the user as "mod_sofia"... is it possible to do this? Thanks and Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 ᐧ -------------- next part -------------- An HTML attachment was scrubbed... URL: From hardikitpl at gmail.com Fri Jun 9 10:08:39 2017 From: hardikitpl at gmail.com (Hardik Patel) Date: Fri, 9 Jun 2017 15:38:39 +0530 Subject: [Freeswitch-users] Fax receive issue with t30 codec In-Reply-To: References: Message-ID: Hi Brian, We have update Freeswitch version but still fax is not working using t30. We are getting below time out related error. 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:496 ============================================================================== 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:508 Fax processing not successful - result (3) Timed out waiting for the first message. 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:513 Remote station id: 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:514 Local station id: SpanDSP Fax Ident 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:515 Pages transferred: 0 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:517 Total fax pages: 0 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:518 Image resolution: 0x0 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:519 Transfer Rate: 14400 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:521 ECM status off 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:522 remote country: 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:523 remote vendor: 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:524 remote model: 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:526 ============================================================================== On Tue, Jun 6, 2017 at 7:30 PM, Brian West wrote: > You'll have to use 1.6.17 if you ever want any faxing to work in all test > cases. > > /b > > On Tue, Jun 6, 2017 at 6:11 AM, Hardik Patel wrote: > >> Hello, >> >> I am using opensips as entry point using dispatcher. opensips( >> 127.0.0.1), i am routing call to freeswitch server (127.0.0.3). >> >> Now I am trying to receive fax, my issue is when i try to send fax in >> softphone(Zoiper) from the log i am seeing that it is sending fax using t30 >> codec. and i am not receiving the fax at destination, is it because of >> codec, should it only work with t38 codec? if that is the issue than how am >> i be able to send the fax using t38 from zoiper? >> >> Here i am attaching the fs log with loglevel 9 and sip trace is also >> enabled. >> >> 127.0.0.2 => carrier/provider IP >> 123456789 => Fax number >> test at gamil.com => Email Address >> 127.0.0.4 =>UI IP >> >> Pastebin link:https://pastebin.freeswitch.org/view/9ec52715 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > > *Brian West* > brian at freeswitch.org > > *Twitter: @FreeSWITCH , @briankwest* > > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > Book a phone call (CST) > > Allison prompts for FreeSWITCH: > > *https://www.gofundme.com/allison-prompts-for-freeswitch* > > > Got Bugs? Report them here ! | Reddit: > /r/freeswitch > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *Skype:*briankwest > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Hardik Patel iNextrix Technologies Pvt Ltd -------------- next part -------------- An HTML attachment was scrubbed... URL: From marcel.haldemann at convercom.ch Fri Jun 9 10:30:51 2017 From: marcel.haldemann at convercom.ch (Marcel Haldemann) Date: Fri, 9 Jun 2017 10:30:51 +0000 Subject: [Freeswitch-users] Removing userpart of contact In-Reply-To: References: Message-ID: U should be able to change it via: sip_contact_uri Not sure wheter u need to add sip: or not. From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David Villasmil Sent: Friday, June 9, 2017 7:48 AM To: FreeSWITCH Users Help Subject: [Freeswitch-users] Removing userpart of contact Hello guys, I ran into a situation where I need the contact to be like: > Meaning I need to remove the username, i've trying doing this but FS adds the user as "mod_sofia"... is it possible to do this? Thanks and Regards, David Villasmil email: david.villasmil.work at gmail.com phone: +34669448337 [https://mailfoogae.appspot.com/t?sender=aZGF2aWQudmlsbGFzbWlsLndvcmtAZ21haWwuY29t&type=zerocontent&guid=4dfa2470-b916-44df-b5c7-3822b6d98d25]ᐧ -------------- next part -------------- An HTML attachment was scrubbed... URL: From asilva at wirelessmundi.com Fri Jun 9 14:53:57 2017 From: asilva at wirelessmundi.com (Antonio Silva) Date: Fri, 9 Jun 2017 16:53:57 +0200 Subject: [Freeswitch-users] Open Source SIP\WebRTC clients compatible with FreeSWITCH In-Reply-To: References: <6FD2F8B5BB72834E9939AEDF9FB802A901E860B047@mbx-01.sysconfig.co.uk> <10c301d236a7$1ea0f8f0$5be2ead0$@freeswitch.org> <10cd01d236a8$425136b0$c6f3a410$@freeswitch.org> Message-ID: <6311b75e-abb2-ad9f-8629-e7d564a379e0@wirelessmundi.com> meanwhile if build app from scratch we can use this: https://github.com/ISBX/apprtc-ios Saludos / Regards / Cumprimentos, António silva On 06/07/2017 11:50 PM, Mundkowsky, Robert wrote: > > Cool, not if they just support getUserMedia! > > Robert Mundkowsky > > *From:*FreeSWITCH-users > [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of > *Guillermo Ruiz Camauer > *Sent:* Wednesday, June 7, 2017 4:18 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Open Source SIP\WebRTC clients > compatible with FreeSWITCH > > Apple announces WebRTC support in iOS11 / Safari: > https://apple.slashdot.org/story/17/06/07/1958242/apple-announces-support-for-webrtc-in-safari-11 > > On Fri, Nov 4, 2016 at 11:32 AM, Ken Rice > wrote: > > No one supports Native WebRTC on iOS at this time except for > people using their own private SDKs that they are not allowing to > get out there… > > Apple does not have webRTC in webkit (Safari) or iOS at this time. > Chrome on iOS is not even really Chrome, its just a wrapper around > the WebKIT APIs and is effectively just safari with a few extra > functions and built to look like chrome. > > *From:*freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] *On Behalf > Of *Chandramouli P > *Sent:* Friday, November 4, 2016 9:29 AM > *To:* FreeSWITCH Users Help > > *Subject:* Re: [Freeswitch-users] Open Source SIP\WebRTC clients > compatible with FreeSWITCH > > Hello Ken, > > We worked with Google native WebRTC on Firefox, Chrome, and Opera > browsers on Windows OS. I recently noticed that Google has added > the WebRTC support for Android, and iOS platforms (webrtc.org > ). Now, we are planning to develop video > calling module using Google native WebRTC on these new platforms. > Can anybody give me the information about my below queries: > > 1) Does Google native WebRTC supports Apple iOS platform (native > mobile app)? > > 2) Does Google native WebRTC supports Apple OS X platform? > > 3) Is it possible to develop video calling module using native > WebRTC on Safari, and Chrome browsers on Apple OS X platform? > > 4) Does Google native WebRTC supports Android platform (native > mobile app)? > > 5) If it supports, I could not find any documentation for Apple > iOS, Apple OS X, and Android platforms specifically. Could you > please send some referral links? > > 6) I could not able to find the referral examples also for Apple > iOS, Apple OS X, and Android platforms specifically. Could you > please send some referral links? > > Please do needful. > > Thank you, > > Chandramouli. > > On Fri, Nov 4, 2016 at 7:54 PM, Ken Rice > wrote: > > Verto Works on pretty much any platform that has native webrtc > support now... unfortunately things like iOS and don’t have > native iOS support yet… > > If you are looking to build something you might contact > consulting at freeswitch.org > and see if you can work with the FSS Team to develop something > > *From:* freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] *On > Behalf Of *Shaun Stokes > *Sent:* Friday, November 4, 2016 8:40 AM > *To:* 'FreeSWITCH Users Help' > > > *Subject:* [Freeswitch-users] Open Source SIP\WebRTC clients > compatible with FreeSWITCH > > Hi All, > > Does anyone have any recommendations on a good open source > SIP\WebRTC client which works on multiple platforms (Windows, > Mac, Linux, Mobiles) to provide presence, voice, video, > instant messaging, screen sharing and file sharing? This must > be capable of integrating with FreeSWITCH for voice and video > (presence via FreeSWITCH would be an advantage). > > Many Thanks, > > Shaun > > > > Shaun Stokes - Infrastructure Analyst > > > > > T : > > > > 01453 700713 > > E : > > > > shaun.stokes at itec-support.co.uk > > > W : > > > > www.itec-support.co.uk > > Registered Address :- ITEC Support, Suite 2 Prospect House, > Bath Road, Stroud, Gloucestershire GL5 3QF > Company No. 06908001 > > > CONFIDENTIALITY NOTICE > This communication and the information it contains are > intended for the person or organisation to which it is > addressed. Its contents are confidential and may be protected > in law. Unauthorised use, copying or disclosure of any of it > may be unlawful. If you are not the intended recipient, please > contact us immediately. > The contents of any attachments in this e-mail may contain > software viruses, which could damage your own computer system. > While ITEC Support has taken every reasonable precaution to > minimise this risk, we cannot accept liability for any damage > which you sustain as a result of software viruses. You should > carry out your own virus checking procedure before opening any > attachment. > > > ______________________________________________________________________ > This message has been checked for all known viruses by > MessageLabs Virus Scanning Service. > ______________________________________________________________________ > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at 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 at 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 at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -- > > Guillermo Ruiz Camauer > > > ------------------------------------------------------------------------ > > This e-mail and any files transmitted with it may contain privileged > or confidential information. It is solely for use by the individual > for whom it is intended, even if addressed incorrectly. If you > received this e-mail in error, please notify the sender; do not > disclose, copy, distribute, or take any action in reliance on the > contents of this information; and delete it from your system. Any > other use of this e-mail is prohibited. > > > Thank you for your compliance. > > ------------------------------------------------------------------------ > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From eastour at 163.com Fri Jun 9 07:02:58 2017 From: eastour at 163.com (chenyzhi) Date: Fri, 9 Jun 2017 15:02:58 +0800 (CST) Subject: [Freeswitch-users] When calling out , freeswitch doesn't get DTMF ? In-Reply-To: <15c898b5d58.2779.b07ebdf329620b8089087c7205b03f01@xbipin.com> References: <15c898b5d58.2779.b07ebdf329620b8089087c7205b03f01@xbipin.com> Message-ID: <310cf71c.8fc3.15c8ba9c4f6.Coremail.eastour@163.com> Hi call 5000 from x-lite , you can hear the IVR voice and if you press dtmf keys , freeswitch can receive the dtmf keys . but , if you enter the command "originate user/1001 5000" , x-lite will ring ,answer it ,you can hear the IVR voice , press some keys ,the freeswitch can NOT receive any dtmf , why? -------------- next part -------------- An HTML attachment was scrubbed... URL: From manpower13.cse at gmail.com Fri Jun 9 15:41:04 2017 From: manpower13.cse at gmail.com (Murugan Pandian) Date: Fri, 9 Jun 2017 21:11:04 +0530 Subject: [Freeswitch-users] Open Source SIP\WebRTC clients compatible with FreeSWITCH In-Reply-To: <6311b75e-abb2-ad9f-8629-e7d564a379e0@wirelessmundi.com> References: <6FD2F8B5BB72834E9939AEDF9FB802A901E860B047@mbx-01.sysconfig.co.uk> <10c301d236a7$1ea0f8f0$5be2ead0$@freeswitch.org> <10cd01d236a8$425136b0$c6f3a410$@freeswitch.org> <6311b75e-abb2-ad9f-8629-e7d564a379e0@wirelessmundi.com> Message-ID: HI , You can try this React-native JSSIP,It support WebRTC i just tested this ,It's working fine with Android https://github.com/telecmi/react-native-chub On Fri, Jun 9, 2017 at 8:23 PM, Antonio Silva wrote: > > meanwhile if build app from scratch we can use this: > > https://github.com/ISBX/apprtc-ios > > > Saludos / Regards / Cumprimentos, > António silva > > On 06/07/2017 11:50 PM, Mundkowsky, Robert wrote: > > Cool, not if they just support getUserMedia! > > > > Robert Mundkowsky > > > > *From:* FreeSWITCH-users [mailto:freeswitch-users- > bounces at lists.freeswitch.org > ] *On Behalf Of *Guillermo > Ruiz Camauer > *Sent:* Wednesday, June 7, 2017 4:18 PM > *To:* FreeSWITCH Users Help > > *Subject:* Re: [Freeswitch-users] Open Source SIP\WebRTC clients > compatible with FreeSWITCH > > > > Apple announces WebRTC support in iOS11 / Safari: > https://apple.slashdot.org/story/17/06/07/1958242/apple- > announces-support-for-webrtc-in-safari-11 > > > > > > > > On Fri, Nov 4, 2016 at 11:32 AM, Ken Rice wrote: > > No one supports Native WebRTC on iOS at this time except for people using > their own private SDKs that they are not allowing to get out there… > > > > Apple does not have webRTC in webkit (Safari) or iOS at this time. Chrome > on iOS is not even really Chrome, its just a wrapper around the WebKIT APIs > and is effectively just safari with a few extra functions and built to look > like chrome. > > > > > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Chandramouli > P > *Sent:* Friday, November 4, 2016 9:29 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Open Source SIP\WebRTC clients > compatible with FreeSWITCH > > > > Hello Ken, > > > > We worked with Google native WebRTC on Firefox, Chrome, and Opera browsers > on Windows OS. I recently noticed that Google has added the WebRTC support > for Android, and iOS platforms (webrtc.org). Now, we are planning to > develop video calling module using Google native WebRTC on these new > platforms. Can anybody give me the information about my below queries: > > > > 1) Does Google native WebRTC supports Apple iOS platform (native mobile > app)? > > 2) Does Google native WebRTC supports Apple OS X platform? > > 3) Is it possible to develop video calling module using native WebRTC on > Safari, and Chrome browsers on Apple OS X platform? > > 4) Does Google native WebRTC supports Android platform (native mobile app)? > > 5) If it supports, I could not find any documentation for Apple iOS, Apple > OS X, and Android platforms specifically. Could you please send some > referral links? > > 6) I could not able to find the referral examples also for Apple iOS, > Apple OS X, and Android platforms specifically. Could you please send some > referral links? > > > > Please do needful. > > > > Thank you, > > Chandramouli. > > > > > > On Fri, Nov 4, 2016 at 7:54 PM, Ken Rice wrote: > > Verto Works on pretty much any platform that has native webrtc support > now... unfortunately things like iOS and don’t have native iOS support yet… > > > > If you are looking to build something you might contact > consulting at freeswitch.org and see if you can work with the FSS Team to > develop something > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Shaun Stokes > *Sent:* Friday, November 4, 2016 8:40 AM > *To:* 'FreeSWITCH Users Help' > *Subject:* [Freeswitch-users] Open Source SIP\WebRTC clients compatible > with FreeSWITCH > > > > Hi All, > > > > Does anyone have any recommendations on a good open source SIP\WebRTC > client which works on multiple platforms (Windows, Mac, Linux, Mobiles) to > provide presence, voice, video, instant messaging, screen sharing and file > sharing? This must be capable of integrating with FreeSWITCH for voice and > video (presence via FreeSWITCH would be an advantage). > > > > Many Thanks, > > Shaun > > Shaun Stokes - Infrastructure Analyst > > > T : > > 01453 700713 > > E : > > shaun.stokes at itec-support.co.uk > > W : > > www.itec-support.co.uk > > Registered Address :- ITEC Support, Suite 2 Prospect House, Bath Road, > Stroud, Gloucestershire GL5 3QF > Company No. 06908001 > > > CONFIDENTIALITY NOTICE > This communication and the information it contains are intended for the > person or organisation to which it is addressed. Its contents are > confidential and may be protected in law. Unauthorised use, copying or > disclosure of any of it may be unlawful. If you are not the intended > recipient, please contact us immediately. > The contents of any attachments in this e-mail may contain software > viruses, which could damage your own computer system. While ITEC Support > has taken every reasonable precaution to minimise this risk, we cannot > accept liability for any damage which you sustain as a result of software > viruses. You should carry out your own virus checking procedure before > opening any attachment. > > > ______________________________________________________________________ > This message has been checked for all known viruses by MessageLabs Virus > Scanning Service. > ______________________________________________________________________ > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > -- > > Guillermo Ruiz Camauer > > ------------------------------ > > This e-mail and any files transmitted with it may contain privileged or > confidential information. It is solely for use by the individual for whom > it is intended, even if addressed incorrectly. If you received this e-mail > in error, please notify the sender; do not disclose, copy, distribute, or > take any action in reliance on the contents of this information; and delete > it from your system. Any other use of this e-mail is prohibited. > > Thank you for your compliance. > ------------------------------ > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From dig1234 at gmail.com Fri Jun 9 15:50:48 2017 From: dig1234 at gmail.com (Daniel Greenwald) Date: Fri, 9 Jun 2017 11:50:48 -0400 Subject: [Freeswitch-users] RPORT still being sent in TCP calls Message-ID: We have noticed that FS is sending RPORT in TCP calls to a gateway. It was reported as fixed in this bug: https://freeswitch.org/jira/browse/FS-6612 We are running: 1.5.15b+git~20150512T053645Z~9eb887af47~64bit I am not sure why RPORT is still being sent. Is this there a config parameter which needs to be set to suppress the RPORT? Or was this change reverted in later versions for some reason. Provider is telling us we should not be sending RPORT in TCP... Any info would be greatly appreciated. Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: From krice at freeswitch.org Fri Jun 9 16:06:55 2017 From: krice at freeswitch.org (Ken Rice) Date: Fri, 9 Jun 2017 11:06:55 -0500 Subject: [Freeswitch-users] Open Source SIP\WebRTC clients compatible with FreeSWITCH In-Reply-To: References: <6FD2F8B5BB72834E9939AEDF9FB802A901E860B047@mbx-01.sysconfig.co.uk> <10c301d236a7$1ea0f8f0$5be2ead0$@freeswitch.org> <10cd01d236a8$425136b0$c6f3a410$@freeswitch.org> Message-ID: <4fc801d2e13a$6b052de0$410f89a0$@freeswitch.org> Yes they just announced this and we’re quite excited about it. I’m personally waiting to see what this means not just for Safari but also for iOS apps themselves. Rest assured the FreeSWITCH team will be testing with Safari. From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Guillermo Ruiz Camauer Sent: Wednesday, June 7, 2017 3:18 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Open Source SIP\WebRTC clients compatible with FreeSWITCH Apple announces WebRTC support in iOS11 / Safari: https://apple.slashdot.org/story/17/06/07/1958242/apple-announces-support-for-webrtc-in-safari-11 On Fri, Nov 4, 2016 at 11:32 AM, Ken Rice > wrote: No one supports Native WebRTC on iOS at this time except for people using their own private SDKs that they are not allowing to get out there… Apple does not have webRTC in webkit (Safari) or iOS at this time. Chrome on iOS is not even really Chrome, its just a wrapper around the WebKIT APIs and is effectively just safari with a few extra functions and built to look like chrome. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org ] On Behalf Of Chandramouli P Sent: Friday, November 4, 2016 9:29 AM To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Open Source SIP\WebRTC clients compatible with FreeSWITCH Hello Ken, We worked with Google native WebRTC on Firefox, Chrome, and Opera browsers on Windows OS. I recently noticed that Google has added the WebRTC support for Android, and iOS platforms (webrtc.org ). Now, we are planning to develop video calling module using Google native WebRTC on these new platforms. Can anybody give me the information about my below queries: 1) Does Google native WebRTC supports Apple iOS platform (native mobile app)? 2) Does Google native WebRTC supports Apple OS X platform? 3) Is it possible to develop video calling module using native WebRTC on Safari, and Chrome browsers on Apple OS X platform? 4) Does Google native WebRTC supports Android platform (native mobile app)? 5) If it supports, I could not find any documentation for Apple iOS, Apple OS X, and Android platforms specifically. Could you please send some referral links? 6) I could not able to find the referral examples also for Apple iOS, Apple OS X, and Android platforms specifically. Could you please send some referral links? Please do needful. Thank you, Chandramouli. On Fri, Nov 4, 2016 at 7:54 PM, Ken Rice > wrote: Verto Works on pretty much any platform that has native webrtc support now... unfortunately things like iOS and don’t have native iOS support yet… If you are looking to build something you might contact consulting at freeswitch.org and see if you can work with the FSS Team to develop something From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org ] On Behalf Of Shaun Stokes Sent: Friday, November 4, 2016 8:40 AM To: 'FreeSWITCH Users Help' > Subject: [Freeswitch-users] Open Source SIP\WebRTC clients compatible with FreeSWITCH Hi All, Does anyone have any recommendations on a good open source SIP\WebRTC client which works on multiple platforms (Windows, Mac, Linux, Mobiles) to provide presence, voice, video, instant messaging, screen sharing and file sharing? This must be capable of integrating with FreeSWITCH for voice and video (presence via FreeSWITCH would be an advantage). Many Thanks, Shaun Shaun Stokes - Infrastructure Analyst T : 01453 700713 E : shaun.stokes at itec-support.co.uk W : www.itec-support.co.uk Registered Address :- ITEC Support, Suite 2 Prospect House, Bath Road, Stroud, Gloucestershire GL5 3QF Company No. 06908001 CONFIDENTIALITY NOTICE This communication and the information it contains are intended for the person or organisation to which it is addressed. Its contents are confidential and may be protected in law. Unauthorised use, copying or disclosure of any of it may be unlawful. If you are not the intended recipient, please contact us immediately. The contents of any attachments in this e-mail may contain software viruses, which could damage your own computer system. While ITEC Support has taken every reasonable precaution to minimise this risk, we cannot accept liability for any damage which you sustain as a result of software viruses. You should carry out your own virus checking procedure before opening any attachment. ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at 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 at 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 at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Guillermo Ruiz Camauer -------------- next part -------------- An HTML attachment was scrubbed... URL: From richardm at cellularity.co.uk Fri Jun 9 17:23:44 2017 From: richardm at cellularity.co.uk (Richard Melville) Date: Fri, 9 Jun 2017 18:23:44 +0100 Subject: [Freeswitch-users] Freeswitch Dependencies Message-ID: On 8 June 2017 at 17:01, Michael Jerris wrote: > > On Jun 8, 2017, at 6:33 AM, Richard Melville > wrote: > > I haven't used or built Freeswitch yet, but I'm getting closer to that > point. I will be building from source. Having looked at the documentation > I can see that Debian is preferred, and the use of distros in general. > However, I don't use distros, but rather build my own systems from scratch. > > I've looked at the Centos page ( https://freeswitch.org/conflue > nce/display/FREESWITCH/CentOS+7+and+RHEL+7 ) and under "Building from > source" there appears to be a list of dependencies. I already have most of > those dependencies installed (other than the codecs) but I have three > questions which maybe somebody can answer. > > The first is: can openssl be substituted with libressl, which I use? > > > Not sure, depends if libressl has the required pieces we need for > dtls-srtp and all the required ciphers required by the browsers for webrtc. > Thanks, I suppose the answer is to give it a try. > > > The second is: "mongo-c-driver-devel" suggests that mongodb is a > dependency of Freeswitch. I've seen no mention of mongodb anywhere in > either the book, or the documentation generally, so why is this listed as a > dependency? > > > Its a module, if you don’t want that module, its not needed. > That's what I thought; thanks for the heads up. > > > The third is: there is no mention of package version numbers anywhere, so > how can I find if there are any issues with particular versions? > > > We don’t test a vast array of different package versions, we do testing > based on the ones for the distros we package for. The versions in Debian 8 > are well tested, other versions are much less well tested or not tested at > all. As for other libs, use the ones in our stash project for dep libs > when not otherwise available. Creating extensive documentation for > building on your own distro would be far more work than even adding support > for a new distro, and we don’t have any plans to create that. I’m happy to > respond to some specific questions, but there are limits to the amount of > time that it makes sense for us to spend on issues like this for a single > person. > Again, I suppose the answer is to just try the build with my versions (most of which are fairly recent -- maybe too recent, but I'll see) and if there are any problems with a particular package then refer back to Debian 8. > > > Any help gratefully received. > > Thanks for your help, Michael. -- Richard Melville Systems Architect cellularity.co.uk stellarsystem.wordpress.com +44 20 33 555 305 +44 7957 836330 -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Fri Jun 9 18:01:54 2017 From: mike at jerris.com (Michael Jerris) Date: Fri, 9 Jun 2017 14:01:54 -0400 Subject: [Freeswitch-users] Freeswitch Dependencies In-Reply-To: References: Message-ID: <7A1E6C8D-009F-45B2-8F02-D3946EC64DA9@jerris.com> > On Jun 9, 2017, at 1:23 PM, Richard Melville wrote: > > On 8 June 2017 at 17:01, Michael Jerris > wrote: > >> On Jun 8, 2017, at 6:33 AM, Richard Melville > wrote: >> >> I haven't used or built Freeswitch yet, but I'm getting closer to that point. I will be building from source. Having looked at the documentation I can see that Debian is preferred, and the use of distros in general. However, I don't use distros, but rather build my own systems from scratch. >> >> I've looked at the Centos page ( https://freeswitch.org/confluence/display/FREESWITCH/CentOS+7+and+RHEL+7 ) and under "Building from source" there appears to be a list of dependencies. I already have most of those dependencies installed (other than the codecs) but I have three questions which maybe somebody can answer. >> >> The first is: can openssl be substituted with libressl, which I use? > > Not sure, depends if libressl has the required pieces we need for dtls-srtp and all the required ciphers required by the browsers for webrtc. > > Thanks, I suppose the answer is to give it a try. > >> >> The second is: "mongo-c-driver-devel" suggests that mongodb is a dependency of Freeswitch. I've seen no mention of mongodb anywhere in either the book, or the documentation generally, so why is this listed as a dependency? > > Its a module, if you don’t want that module, its not needed. > > That's what I thought; thanks for the heads up. > >> >> The third is: there is no mention of package version numbers anywhere, so how can I find if there are any issues with particular versions? > > We don’t test a vast array of different package versions, we do testing based on the ones for the distros we package for. The versions in Debian 8 are well tested, other versions are much less well tested or not tested at all. As for other libs, use the ones in our stash project for dep libs when not otherwise available. Creating extensive documentation for building on your own distro would be far more work than even adding support for a new distro, and we don’t have any plans to create that. I’m happy to respond to some specific questions, but there are limits to the amount of time that it makes sense for us to spend on issues like this for a single person. > > Again, I suppose the answer is to just try the build with my versions (most of which are fairly recent -- maybe too recent, but I'll see) and if there are any problems with a particular package then refer back to Debian 8. No problem. The easiest way to see the module dep chain would be to look at this file: https://freeswitch.org/stash/projects/FS/repos/freeswitch/browse/debian/control-modules > >> >> Any help gratefully received. >> > > Thanks for your help, Michael. > > -- > Richard Melville > Systems Architect > cellularity.co.uk > stellarsystem.wordpress.com > +44 20 33 555 305 > +44 7957 836330 > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From richardm at cellularity.co.uk Fri Jun 9 18:22:31 2017 From: richardm at cellularity.co.uk (Richard Melville) Date: Fri, 9 Jun 2017 19:22:31 +0100 Subject: [Freeswitch-users] Freeswitch Dependencies Message-ID: On 8 June 2017 at 22:34, Ken Rice wrote: > > And as usual I got interrupted and clicked send before finishing a > thought… (sorry about that but it helps demonstrate our time constraints) > > > > But following up my earlier comments > > > > > > Now that being said you can review the freeswitch.spec file in the root of > the source tree or the Debian packaging files in the debian dir also in the > source tree to get a list of build deps and if min versions are required > what those are. > > > > Things without version numbering in those files are that way due to the > current versions in the appropriate Distros being sufficient and Distros > typically do not change APIs once theyhave released a version (or on minor > versions upgrades)_ > > Thanks Ken. In reply to this, and your earlier email, that's really useful info. I'm not looking for one-off support, but just to be pointed in roughly the right direction. Regarding LibreSSL I guess that it's just a case of trying it out. Thanks again. -- Richard Melville Systems Architect cellularity.co.uk stellarsystem.wordpress.com +44 20 33 555 305 +44 7957 836330 -------------- next part -------------- An HTML attachment was scrubbed... URL: From richardm at cellularity.co.uk Fri Jun 9 18:27:59 2017 From: richardm at cellularity.co.uk (Richard Melville) Date: Fri, 9 Jun 2017 19:27:59 +0100 Subject: [Freeswitch-users] Freeswitch Dependencies Message-ID: On 9 June 2017 at 19:01, Michael Jerris wrote: > > On Jun 9, 2017, at 1:23 PM, Richard Melville > wrote: > > On 8 June 2017 at 17:01, Michael Jerris wrote: > >> >> On Jun 8, 2017, at 6:33 AM, Richard Melville >> wrote: >> >> I haven't used or built Freeswitch yet, but I'm getting closer to that >> point. I will be building from source. Having looked at the documentation >> I can see that Debian is preferred, and the use of distros in general. >> However, I don't use distros, but rather build my own systems from scratch. >> >> I've looked at the Centos page ( https://freeswitch.org/conflue >> nce/display/FREESWITCH/CentOS+7+and+RHEL+7 ) and under "Building from >> source" there appears to be a list of dependencies. I already have most of >> those dependencies installed (other than the codecs) but I have three >> questions which maybe somebody can answer. >> >> The first is: can openssl be substituted with libressl, which I use? >> >> >> Not sure, depends if libressl has the required pieces we need for >> dtls-srtp and all the required ciphers required by the browsers for webrtc. >> > > Thanks, I suppose the answer is to give it a try. > >> >> >> The second is: "mongo-c-driver-devel" suggests that mongodb is a >> dependency of Freeswitch. I've seen no mention of mongodb anywhere in >> either the book, or the documentation generally, so why is this listed as a >> dependency? >> >> >> Its a module, if you don’t want that module, its not needed. >> > > That's what I thought; thanks for the heads up. > >> >> >> The third is: there is no mention of package version numbers anywhere, so >> how can I find if there are any issues with particular versions? >> >> >> We don’t test a vast array of different package versions, we do testing >> based on the ones for the distros we package for. The versions in Debian 8 >> are well tested, other versions are much less well tested or not tested at >> all. As for other libs, use the ones in our stash project for dep libs >> when not otherwise available. Creating extensive documentation for >> building on your own distro would be far more work than even adding support >> for a new distro, and we don’t have any plans to create that. I’m happy to >> respond to some specific questions, but there are limits to the amount of >> time that it makes sense for us to spend on issues like this for a single >> person. >> > > Again, I suppose the answer is to just try the build with my versions > (most of which are fairly recent -- maybe too recent, but I'll see) and if > there are any problems with a particular package then refer back to Debian > 8. > > > No problem. The easiest way to see the module dep chain would be to look > at this file: > > https://freeswitch.org/stash/projects/FS/repos/freeswitch/ > browse/debian/control-modules > > Excellent, thanks for that. -- Richard Melville Systems Architect cellularity.co.uk stellarsystem.wordpress.com +44 20 33 555 305 +44 7957 836330 -------------- next part -------------- An HTML attachment was scrubbed... URL: From anthony.minessale at gmail.com Fri Jun 9 18:56:47 2017 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 9 Jun 2017 13:56:47 -0500 Subject: [Freeswitch-users] RPORT still being sent in TCP calls In-Reply-To: References: Message-ID: Odd number releases are not stable releases they are dev releases. The version you quoted is from may 2015 2 years ago. The latest stable release is 1.6.17 and latest build release changes every day. On Fri, Jun 9, 2017 at 10:50 AM, Daniel Greenwald wrote: > We have noticed that FS is sending RPORT in TCP calls to a gateway. It was > reported as fixed in this bug: > https://freeswitch.org/jira/browse/FS-6612 > > We are running: > 1.5.15b+git~20150512T053645Z~9eb887af47~64bit > > I am not sure why RPORT is still being sent. Is this there a config > parameter which needs to be set to suppress the RPORT? Or was this change > reverted in later versions for some reason. Provider is telling us we > should not be sending RPORT in TCP... > > Any info would be greatly appreciated. > > Thanks! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at 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 ♬ @anthmfs ♬ @FreeSWITCH ♬ ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ * ClueCon Weekly Development Call ☎ sip:888 at conference.freeswitch.org ☎ +19193869900 https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA -------------- next part -------------- An HTML attachment was scrubbed... URL: From tculjaga at gmail.com Fri Jun 9 20:42:24 2017 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Fri, 9 Jun 2017 22:42:24 +0200 Subject: [Freeswitch-users] Fax receive issue with t30 codec In-Reply-To: References: Message-ID: hi, whats new on faxing in 1.6.17 ? T. On 6 June 2017 at 16:00, Brian West wrote: > You'll have to use 1.6.17 if you ever want any faxing to work in all test > cases. > > /b > > On Tue, Jun 6, 2017 at 6:11 AM, Hardik Patel wrote: > >> Hello, >> >> I am using opensips as entry point using dispatcher. opensips( >> 127.0.0.1), i am routing call to freeswitch server (127.0.0.3). >> >> Now I am trying to receive fax, my issue is when i try to send fax in >> softphone(Zoiper) from the log i am seeing that it is sending fax using t30 >> codec. and i am not receiving the fax at destination, is it because of >> codec, should it only work with t38 codec? if that is the issue than how am >> i be able to send the fax using t38 from zoiper? >> >> Here i am attaching the fs log with loglevel 9 and sip trace is also >> enabled. >> >> 127.0.0.2 => carrier/provider IP >> 123456789 => Fax number >> test at gamil.com => Email Address >> 127.0.0.4 =>UI IP >> >> Pastebin link:https://pastebin.freeswitch.org/view/9ec52715 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > > *Brian West* > brian at freeswitch.org > > *Twitter: @FreeSWITCH , @briankwest* > > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > Book a phone call (CST) > > Allison prompts for FreeSWITCH: > > *https://www.gofundme.com/allison-prompts-for-freeswitch* > > > Got Bugs? Report them here ! | Reddit: > /r/freeswitch > > *T:*+19184209001 <+1%20918-420-9001> | *F:*+19184209002 > <+1%20918-420-9002> | *M:*+1918424WEST (9378) > *Skype:*briankwest > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian at freeswitch.org Fri Jun 9 21:12:51 2017 From: brian at freeswitch.org (Brian West) Date: Fri, 9 Jun 2017 16:12:51 -0500 Subject: [Freeswitch-users] Fax receive issue with t30 codec In-Reply-To: References: Message-ID: Your log snip doesn't really help, I know without a single doubt faxing works fine. So what are you doing and how are you doing it? On Fri, Jun 9, 2017 at 3:42 PM, Tihomir Culjaga wrote: > hi, whats new on faxing in 1.6.17 ? > > T. > > On 6 June 2017 at 16:00, Brian West wrote: > >> You'll have to use 1.6.17 if you ever want any faxing to work in all test >> cases. >> >> /b >> >> On Tue, Jun 6, 2017 at 6:11 AM, Hardik Patel >> wrote: >> >>> Hello, >>> >>> I am using opensips as entry point using dispatcher. opensips( >>> 127.0.0.1), i am routing call to freeswitch server (127.0.0.3). >>> >>> Now I am trying to receive fax, my issue is when i try to send fax in >>> softphone(Zoiper) from the log i am seeing that it is sending fax using t30 >>> codec. and i am not receiving the fax at destination, is it because of >>> codec, should it only work with t38 codec? if that is the issue than how am >>> i be able to send the fax using t38 from zoiper? >>> >>> Here i am attaching the fs log with loglevel 9 and sip trace is also >>> enabled. >>> >>> 127.0.0.2 => carrier/provider IP >>> 123456789 => Fax number >>> test at gamil.com => Email Address >>> 127.0.0.4 =>UI IP >>> >>> Pastebin link:https://pastebin.freeswitch.org/view/9ec52715 >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at 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 at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> >> *Brian West* >> brian at freeswitch.org >> >> *Twitter: @FreeSWITCH , @briankwest* >> >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> Book a phone call (CST) >> >> Allison prompts for FreeSWITCH: >> >> *https://www.gofundme.com/allison-prompts-for-freeswitch* >> >> >> Got Bugs? Report them here ! | Reddit: >> /r/freeswitch >> >> *T:*+19184209001 <+1%20918-420-9001> | *F:*+19184209002 >> <+1%20918-420-9002> | *M:*+1918424WEST (9378) >> *Skype:*briankwest >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Book a phone call (CST) Allison prompts for FreeSWITCH: *https://www.gofundme.com/allison-prompts-for-freeswitch* Got Bugs? Report them here ! | Reddit: /r/freeswitch *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: From chad at apartmentlines.com Fri Jun 9 22:18:38 2017 From: chad at apartmentlines.com (Chad Phillips) Date: Fri, 9 Jun 2017 15:18:38 -0700 Subject: [Freeswitch-users] Open Source SIP\WebRTC clients compatible with FreeSWITCH In-Reply-To: <4fc801d2e13a$6b052de0$410f89a0$@freeswitch.org> References: <6FD2F8B5BB72834E9939AEDF9FB802A901E860B047@mbx-01.sysconfig.co.uk> <10c301d236a7$1ea0f8f0$5be2ead0$@freeswitch.org> <10cd01d236a8$425136b0$c6f3a410$@freeswitch.org> <4fc801d2e13a$6b052de0$410f89a0$@freeswitch.org> Message-ID: Looks like the Janus devs got Safari working pretty easily: https://twitter.com/elminiero/status/873178028683255808 On Fri, Jun 9, 2017 at 9:06 AM, Ken Rice wrote: > Yes they just announced this and we’re quite excited about it. I’m > personally waiting to see what this means not just for Safari but also for > iOS apps themselves. > > > > Rest assured the FreeSWITCH team will be testing with Safari. > > > > *From:* FreeSWITCH-users [mailto:freeswitch-users- > bounces at lists.freeswitch.org] *On Behalf Of *Guillermo Ruiz Camauer > *Sent:* Wednesday, June 7, 2017 3:18 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Open Source SIP\WebRTC clients > compatible with FreeSWITCH > > > > Apple announces WebRTC support in iOS11 / Safari: > https://apple.slashdot.org/story/17/06/07/1958242/apple- > announces-support-for-webrtc-in-safari-11 > > > > > > > > On Fri, Nov 4, 2016 at 11:32 AM, Ken Rice wrote: > > No one supports Native WebRTC on iOS at this time except for people using > their own private SDKs that they are not allowing to get out there… > > > > Apple does not have webRTC in webkit (Safari) or iOS at this time. Chrome > on iOS is not even really Chrome, its just a wrapper around the WebKIT APIs > and is effectively just safari with a few extra functions and built to look > like chrome. > > > > > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Chandramouli > P > *Sent:* Friday, November 4, 2016 9:29 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Open Source SIP\WebRTC clients > compatible with FreeSWITCH > > > > Hello Ken, > > > > We worked with Google native WebRTC on Firefox, Chrome, and Opera browsers > on Windows OS. I recently noticed that Google has added the WebRTC support > for Android, and iOS platforms (webrtc.org). Now, we are planning to > develop video calling module using Google native WebRTC on these new > platforms. Can anybody give me the information about my below queries: > > > > 1) Does Google native WebRTC supports Apple iOS platform (native mobile > app)? > > 2) Does Google native WebRTC supports Apple OS X platform? > > 3) Is it possible to develop video calling module using native WebRTC on > Safari, and Chrome browsers on Apple OS X platform? > > 4) Does Google native WebRTC supports Android platform (native mobile app)? > > 5) If it supports, I could not find any documentation for Apple iOS, Apple > OS X, and Android platforms specifically. Could you please send some > referral links? > > 6) I could not able to find the referral examples also for Apple iOS, > Apple OS X, and Android platforms specifically. Could you please send some > referral links? > > > > Please do needful. > > > > Thank you, > > Chandramouli. > > > > > > On Fri, Nov 4, 2016 at 7:54 PM, Ken Rice wrote: > > Verto Works on pretty much any platform that has native webrtc support > now... unfortunately things like iOS and don’t have native iOS support yet… > > > > If you are looking to build something you might contact > consulting at freeswitch.org and see if you can work with the FSS Team to > develop something > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Shaun Stokes > *Sent:* Friday, November 4, 2016 8:40 AM > *To:* 'FreeSWITCH Users Help' > *Subject:* [Freeswitch-users] Open Source SIP\WebRTC clients compatible > with FreeSWITCH > > > > Hi All, > > > > Does anyone have any recommendations on a good open source SIP\WebRTC > client which works on multiple platforms (Windows, Mac, Linux, Mobiles) to > provide presence, voice, video, instant messaging, screen sharing and file > sharing? This must be capable of integrating with FreeSWITCH for voice and > video (presence via FreeSWITCH would be an advantage). > > > > Many Thanks, > > Shaun > > Shaun Stokes - Infrastructure Analyst > > T : > > 01453 700713 > > E : > > shaun.stokes at itec-support.co.uk > > W : > > www.itec-support.co.uk > > Registered Address :- ITEC Support, Suite 2 Prospect House, Bath Road, > Stroud, Gloucestershire GL5 3QF > Company No. 06908001 > > > CONFIDENTIALITY NOTICE > This communication and the information it contains are intended for the > person or organisation to which it is addressed. Its contents are > confidential and may be protected in law. Unauthorised use, copying or > disclosure of any of it may be unlawful. If you are not the intended > recipient, please contact us immediately. > The contents of any attachments in this e-mail may contain software > viruses, which could damage your own computer system. While ITEC Support > has taken every reasonable precaution to minimise this risk, we cannot > accept liability for any damage which you sustain as a result of software > viruses. You should carry out your own virus checking procedure before > opening any attachment. > > > ______________________________________________________________________ > This message has been checked for all known viruses by MessageLabs Virus > Scanning Service. > ______________________________________________________________________ > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > -- > > Guillermo Ruiz Camauer > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil at gmail.com Fri Jun 9 22:48:18 2017 From: david.villasmil at gmail.com (David Villasmil Govea) Date: Fri, 09 Jun 2017 22:48:18 +0000 Subject: [Freeswitch-users] Removing userpart of contact In-Reply-To: References: Message-ID: Hello, thanks for your help, but i had already tried that, and it doesn't do what i need, the contact still is like "mod_sofia at 1.2.3.4" I was looking at the source, and it seems to me, if the contact begins with "@" which i assume is added somewhere i couldn't find, then fs adds the "mod_sofia"... Any thoughts? Regards, David On Fri, Jun 9, 2017 at 6:31 AM Marcel Haldemann < marcel.haldemann at convercom.ch> wrote: > U should be able to change it via: > > > > sip_contact_uri > > > > > > > > Not sure wheter u need to add sip: or not. > > > > *From:* FreeSWITCH-users [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *David > Villasmil > *Sent:* Friday, June 9, 2017 7:48 AM > *To:* FreeSWITCH Users Help > *Subject:* [Freeswitch-users] Removing userpart of contact > > > > Hello guys, > > > > I ran into a situation where I need the contact to be like: > > > > > > Meaning I need to remove the username, i've trying doing this but FS adds > the user as "mod_sofia"... is it possible to do this? > > > > > > Thanks and Regards, > > > > David Villasmil > > email: david.villasmil.work at gmail.com > > phone: +34669448337 > > ᐧ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Fri Jun 9 22:49:34 2017 From: david.villasmil.work at gmail.com (David Villasmil) Date: Fri, 09 Jun 2017 22:49:34 +0000 Subject: [Freeswitch-users] When calling out , freeswitch doesn't get DTMF ? In-Reply-To: <310cf71c.8fc3.15c8ba9c4f6.Coremail.eastour@163.com> References: <15c898b5d58.2779.b07ebdf329620b8089087c7205b03f01@xbipin.com> <310cf71c.8fc3.15c8ba9c4f6.Coremail.eastour@163.com> Message-ID: Have you looked at the log? Bump the logging up and see what shows up... what you're seeing is very weird David On Fri, Jun 9, 2017 at 11:18 AM chenyzhi wrote: > Hi > > call 5000 from x-lite , you can hear the IVR voice and if you press dtmf > keys , freeswitch can receive the dtmf keys . > > but , if you enter the command "originate user/1001 5000" , x-lite will > ring ,answer it ,you can hear the IVR voice , press some keys ,the > freeswitch can NOT receive any dtmf , > > why? > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From hardikitpl at gmail.com Sat Jun 10 04:16:51 2017 From: hardikitpl at gmail.com (Hardik Patel) Date: Sat, 10 Jun 2017 09:46:51 +0530 Subject: [Freeswitch-users] Fax receive issue with t30 codec In-Reply-To: References: Message-ID: Hi Brian, Thanks for the support. We are testing receive fax functionality using real fax machine and here i have listed the model which we are using to send fax. the models of real fax machines that we have used are group 3 CCITT / ITU, they are the following: 1 ) konica minolta bizhub-c220 2 ) HP Officejet 4500 3 ) HP Officejet G85 >From above list one of our machine is sending fax without T38 support and we got the failure with same error which have posted on bug but if we use T38 support then it works fine for us. *ERROR :* 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:508 Fax processing not successful - result (3) Timed out waiting for the first message. 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:513 Remote station id: 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:514 Local station id: SpanDSP Fax Ident 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:515 Pages transferred: 0 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:517 Total fax pages: 0 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:518 Image resolution: 0x0 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:519 Transfer Rate: 14400 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:521 ECM status off 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:522 remote country: 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:523 remote vendor: 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:524 remote model: 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:526 ============================== On Sat, Jun 10, 2017 at 2:42 AM, Brian West wrote: > Your log snip doesn't really help, I know without a single doubt faxing > works fine. So what are you doing and how are you doing it? > > On Fri, Jun 9, 2017 at 3:42 PM, Tihomir Culjaga > wrote: > >> hi, whats new on faxing in 1.6.17 ? >> >> T. >> >> On 6 June 2017 at 16:00, Brian West wrote: >> >>> You'll have to use 1.6.17 if you ever want any faxing to work in all >>> test cases. >>> >>> /b >>> >>> On Tue, Jun 6, 2017 at 6:11 AM, Hardik Patel >>> wrote: >>> >>>> Hello, >>>> >>>> I am using opensips as entry point using dispatcher. opensips( >>>> 127.0.0.1), i am routing call to freeswitch server (127.0.0.3). >>>> >>>> Now I am trying to receive fax, my issue is when i try to send fax in >>>> softphone(Zoiper) from the log i am seeing that it is sending fax using t30 >>>> codec. and i am not receiving the fax at destination, is it because of >>>> codec, should it only work with t38 codec? if that is the issue than how am >>>> i be able to send the fax using t38 from zoiper? >>>> >>>> Here i am attaching the fs log with loglevel 9 and sip trace is also >>>> enabled. >>>> >>>> 127.0.0.2 => carrier/provider IP >>>> 123456789 => Fax number >>>> test at gamil.com => Email Address >>>> 127.0.0.4 =>UI IP >>>> >>>> Pastebin link:https://pastebin.freeswitch.org/view/9ec52715 >>>> >>>> ____________________________________________________________ >>>> _____________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at 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 at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>> switch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> >>> -- >>> >>> *Brian West* >>> brian at freeswitch.org >>> >>> *Twitter: @FreeSWITCH , @briankwest* >>> >>> http://www.freeswitchbook.com >>> http://www.freeswitchcookbook.com >>> >>> Book a phone call (CST) >>> >>> Allison prompts for FreeSWITCH: >>> >>> *https://www.gofundme.com/allison-prompts-for-freeswitch* >>> >>> >>> Got Bugs? Report them here ! | Reddit: >>> /r/freeswitch >>> >>> *T:*+19184209001 <+1%20918-420-9001> | *F:*+19184209002 >>> <+1%20918-420-9002> | *M:*+1918424WEST (9378) >>> *Skype:*briankwest >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at 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 at 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 at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > > *Brian West* > brian at freeswitch.org > > *Twitter: @FreeSWITCH , @briankwest* > > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > Book a phone call (CST) > > Allison prompts for FreeSWITCH: > > *https://www.gofundme.com/allison-prompts-for-freeswitch* > > > Got Bugs? Report them here ! | Reddit: > /r/freeswitch > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *Skype:*briankwest > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Hardik Patel iNextrix Technologies Pvt Ltd -------------- next part -------------- An HTML attachment was scrubbed... URL: From john at industromatic.com Fri Jun 9 23:33:26 2017 From: john at industromatic.com (John Griessen) Date: Fri, 9 Jun 2017 18:33:26 -0500 Subject: [Freeswitch-users] latency and performance Message-ID: <44eb01b6-b9d6-fad6-fd40-43447167b9aa@industromatic.com> Say your office is in Austin. Can you run a freeswitch server in a Dallas datacenter with a ping time of 15ms and get useful performance using a voip phone in Austin? I've read that running in a VM ruins/randomizes some of the timing the kernel needs to keep voice packets coming well... Is there a way to measure if that is happening too much? -- John Griessen From caleb at bclife.biz Sat Jun 10 04:14:07 2017 From: caleb at bclife.biz (Caleb Bartholomew) Date: Fri, 9 Jun 2017 22:14:07 -0600 Subject: [Freeswitch-users] Multichannel Transcdription Message-ID: <6E3BD1FE-212C-4005-9A4B-8C19DE18758C@bclife.biz> Hi all, I am currently trying to find a solution inside of Freeswitch that will allow me to either record each participant of a conference room separately so it can later be processed for transcription. I’ve looked into a lot of different methods that might be able to do this but I’m not sure any of them are quite what I need. I stumbled across mod_vlc which does hint at this ability but I’m not sure what exactly a “raw” conference stream is or its format. Any help is greatly appreciated. Thanks, Caleb -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 842 bytes Desc: Message signed with OpenPGP URL: From rbetancor at gmail.com Sat Jun 10 10:30:43 2017 From: rbetancor at gmail.com (=?UTF-8?Q?Ra=C3=BAl_Alexis_Betancor_Santana?=) Date: Sat, 10 Jun 2017 11:30:43 +0100 Subject: [Freeswitch-users] latency and performance In-Reply-To: <44eb01b6-b9d6-fad6-fd40-43447167b9aa@industromatic.com> References: <44eb01b6-b9d6-fad6-fd40-43447167b9aa@industromatic.com> Message-ID: You could have your vpbx on Kracovia and your phones on Patagonia, 250ms of RTT and still run your ip voice service smooth. 2017-06-10 0:33 GMT+01:00 John Griessen : > Say your office is in Austin. Can you run a freeswitch server in a Dallas > datacenter with a ping time of 15ms and get useful performance using a voip > phone in Austin? I've read that running in a VM ruins/randomizes some of > the timing the kernel needs > to keep voice packets coming well... Is there a way to measure if that is > happening too much? > > -- > John Griessen > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Sat Jun 10 20:40:44 2017 From: david.villasmil.work at gmail.com (David Villasmil) Date: Sat, 10 Jun 2017 20:40:44 +0000 Subject: [Freeswitch-users] latency and performance In-Reply-To: References: <44eb01b6-b9d6-fad6-fd40-43447167b9aa@industromatic.com> Message-ID: Nothing like testing it yourself. It should be ok as long as you don't put heavy load on it. On Sat, Jun 10, 2017 at 6:31 AM Raúl Alexis Betancor Santana < rbetancor at gmail.com> wrote: > You could have your vpbx on Kracovia and your phones on Patagonia, 250ms > of RTT and still run your ip voice service smooth. > > 2017-06-10 0:33 GMT+01:00 John Griessen : > >> Say your office is in Austin. Can you run a freeswitch server in a >> Dallas datacenter with a ping time of 15ms and get useful performance using >> a voip phone in Austin? I've read that running in a VM ruins/randomizes >> some of the timing the kernel needs >> to keep voice packets coming well... Is there a way to measure if that >> is happening too much? >> >> -- >> John Griessen >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Sun Jun 11 01:39:04 2017 From: david.villasmil.work at gmail.com (David Villasmil) Date: Sun, 11 Jun 2017 01:39:04 +0000 Subject: [Freeswitch-users] Removing userpart of contact In-Reply-To: References: Message-ID: Any thoughts on this, guys? On Fri, Jun 9, 2017 at 6:49 PM David Villasmil Govea < david.villasmil at gmail.com> wrote: > Hello, > > thanks for your help, but i had already tried that, and it doesn't do what > i need, the contact still is like "mod_sofia at 1.2.3.4" > > I was looking at the source, and it seems to me, if the contact begins > with "@" which i assume is added somewhere i couldn't find, then fs adds > the "mod_sofia"... > > Any thoughts? > > Regards, > > David > > > On Fri, Jun 9, 2017 at 6:31 AM Marcel Haldemann < > marcel.haldemann at convercom.ch> wrote: > >> U should be able to change it via: >> >> >> >> sip_contact_uri >> >> >> >> >> >> >> >> Not sure wheter u need to add sip: or not. >> >> >> >> *From:* FreeSWITCH-users [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *David >> Villasmil >> *Sent:* Friday, June 9, 2017 7:48 AM >> *To:* FreeSWITCH Users Help >> *Subject:* [Freeswitch-users] Removing userpart of contact >> >> >> >> Hello guys, >> >> >> >> I ran into a situation where I need the contact to be like: >> >> >> >> >> >> Meaning I need to remove the username, i've trying doing this but FS adds >> the user as "mod_sofia"... is it possible to do this? >> >> >> >> >> >> Thanks and Regards, >> >> >> >> David Villasmil >> >> email: david.villasmil.work at gmail.com >> >> phone: +34669448337 >> >> ᐧ >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From ksrigo at gmail.com Sun Jun 11 10:21:03 2017 From: ksrigo at gmail.com (Srigo Kana) Date: Sun, 11 Jun 2017 12:21:03 +0200 Subject: [Freeswitch-users] Removing userpart of contact In-Reply-To: References: Message-ID: <8DF8A1B6-5E7F-412E-9CB4-70F80578A2EC@gmail.com> Hi, Cld u post ur dialplan where you are doing your bridge? Srigo Sent from my iPhone > On 11 Jun 2017, at 03:39, David Villasmil wrote: > > Any thoughts on this, guys? >> On Fri, Jun 9, 2017 at 6:49 PM David Villasmil Govea wrote: >> Hello, >> >> thanks for your help, but i had already tried that, and it doesn't do what i need, the contact still is like "mod_sofia at 1.2.3.4" >> >> I was looking at the source, and it seems to me, if the contact begins with "@" which i assume is added somewhere i couldn't find, then fs adds the "mod_sofia"... >> >> Any thoughts? >> >> Regards, >> >> David >> >> >>> On Fri, Jun 9, 2017 at 6:31 AM Marcel Haldemann wrote: >>> U should be able to change it via: >>> >>> >>> >>> sip_contact_uri >>> >>> >>> >>> >>> >>> >>> >>> Not sure wheter u need to add sip: or not. >>> >>> >>> >>> From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of David Villasmil >>> Sent: Friday, June 9, 2017 7:48 AM >>> To: FreeSWITCH Users Help >>> Subject: [Freeswitch-users] Removing userpart of contact >>> >>> >>> >>> Hello guys, >>> >>> >>> >>> I ran into a situation where I need the contact to be like: >>> >>> >>> >>> >>> >>> Meaning I need to remove the username, i've trying doing this but FS adds the user as "mod_sofia"... is it possible to do this? >>> >>> >>> >>> >>> >>> Thanks and Regards, >>> >>> >>> >>> David Villasmil >>> >>> email: david.villasmil.work at gmail.com >>> >>> phone: +34669448337 >>> >>> ᐧ >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at 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 at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From ssinyagin at gmail.com Sun Jun 11 14:14:42 2017 From: ssinyagin at gmail.com (Stanislav Sinyagin) Date: Sun, 11 Jun 2017 16:14:42 +0200 Subject: [Freeswitch-users] Collection of notification sounds Message-ID: here's a new collection of notification sounds for your telephony projects: https://github.com/voxserv/dzwin enjoy. There are files already converted for various bitrates. You can download the whole lot from "Releases" tab. From gmaruzz at gmail.com Sun Jun 11 14:54:32 2017 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Sun, 11 Jun 2017 16:54:32 +0200 Subject: [Freeswitch-users] Collection of notification sounds In-Reply-To: References: Message-ID: Very nice, thanks! On 11 June 2017 at 16:14, Stanislav Sinyagin wrote: > here's a new collection of notification sounds for your telephony projects: > https://github.com/voxserv/dzwin > > enjoy. There are files already converted for various bitrates. You can > download the whole lot from "Releases" tab. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at 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 OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil at gmail.com Sun Jun 11 16:08:10 2017 From: david.villasmil at gmail.com (David Villasmil Govea) Date: Sun, 11 Jun 2017 12:08:10 -0400 Subject: [Freeswitch-users] Removing userpart of contact In-Reply-To: <8DF8A1B6-5E7F-412E-9CB4-70F80578A2EC@gmail.com> References: <8DF8A1B6-5E7F-412E-9CB4-70F80578A2EC@gmail.com> Message-ID: hello, here's the dialplan: I've tried several ways, but it seems everytime I set the contact to be "sip:1.2.3.4:5060", fs adds a "mod_sofia" as the user part... Thanks for any help David ᐧ 2017-06-11 6:21 GMT-04:00 Srigo Kana : > Hi, > > Cld u post ur dialplan where you are doing your bridge? > > Srigo > > Sent from my iPhone > > On 11 Jun 2017, at 03:39, David Villasmil > wrote: > > Any thoughts on this, guys? > On Fri, Jun 9, 2017 at 6:49 PM David Villasmil Govea < > david.villasmil at gmail.com> wrote: > >> Hello, >> >> thanks for your help, but i had already tried that, and it doesn't do >> what i need, the contact still is like "mod_sofia at 1.2.3.4" >> >> I was looking at the source, and it seems to me, if the contact begins >> with "@" which i assume is added somewhere i couldn't find, then fs adds >> the "mod_sofia"... >> >> Any thoughts? >> >> Regards, >> >> David >> >> >> On Fri, Jun 9, 2017 at 6:31 AM Marcel Haldemann < >> marcel.haldemann at convercom.ch> wrote: >> >>> U should be able to change it via: >>> >>> >>> >>> sip_contact_uri >>> >>> >>> >>> >>> >>> >>> >>> Not sure wheter u need to add sip: or not. >>> >>> >>> >>> *From:* FreeSWITCH-users [mailto:freeswitch-users- >>> bounces at lists.freeswitch.org] *On Behalf Of *David Villasmil >>> *Sent:* Friday, June 9, 2017 7:48 AM >>> *To:* FreeSWITCH Users Help >>> *Subject:* [Freeswitch-users] Removing userpart of contact >>> >>> >>> >>> Hello guys, >>> >>> >>> >>> I ran into a situation where I need the contact to be like: >>> >>> >>> >>> >>> >>> Meaning I need to remove the username, i've trying doing this but FS >>> adds the user as "mod_sofia"... is it possible to do this? >>> >>> >>> >>> >>> >>> Thanks and Regards, >>> >>> >>> >>> David Villasmil >>> >>> email: david.villasmil.work at gmail.com >>> >>> phone: +34669448337 >>> >>> ᐧ >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at 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 at 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 at 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 at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- DVG -- Imagination is more important than knowledge Albert Einstein -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Sun Jun 11 16:43:18 2017 From: david.villasmil.work at gmail.com (David Villasmil) Date: Sun, 11 Jun 2017 16:43:18 +0000 Subject: [Freeswitch-users] Collection of notification sounds In-Reply-To: References: Message-ID: Thanks! On Sun, Jun 11, 2017 at 10:56 AM Giovanni Maruzzelli wrote: > Very nice, thanks! > > > > On 11 June 2017 at 16:14, Stanislav Sinyagin wrote: > >> here's a new collection of notification sounds for your telephony >> projects: >> https://github.com/voxserv/dzwin >> >> enjoy. There are files already converted for various bitrates. You can >> download the whole lot from "Releases" tab. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at 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 > OpenTelecom.IT > cell: +39 347 266 56 18 > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From eastour at 163.com Mon Jun 12 00:32:53 2017 From: eastour at 163.com (chenyzhi) Date: Mon, 12 Jun 2017 08:32:53 +0800 (CST) Subject: [Freeswitch-users] When calling out , freeswitch doesn't get DTMF ? In-Reply-To: References: <15c898b5d58.2779.b07ebdf329620b8089087c7205b03f01@xbipin.com> <310cf71c.8fc3.15c8ba9c4f6.Coremail.eastour@163.com> Message-ID: <2a18422f.ef8.15c99b7b9bb.Coremail.eastour@163.com> I have read the logs ,but I didn't find any difference. Please make a test to see if this happens in your box. At 2017-06-10 06:49:34, "David Villasmil" wrote: Have you looked at the log? Bump the logging up and see what shows up... what you're seeing is very weird David On Fri, Jun 9, 2017 at 11:18 AM chenyzhi wrote: Hi call 5000 from x-lite , you can hear the IVR voice and if you press dtmf keys , freeswitch can receive the dtmf keys . but , if you enter the command "originate user/1001 5000" , x-lite will ring ,answer it ,you can hear the IVR voice , press some keys ,the freeswitch can NOT receive any dtmf , why? _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From gmaruzz at gmail.com Mon Jun 12 09:37:49 2017 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Mon, 12 Jun 2017 11:37:49 +0200 Subject: [Freeswitch-users] When calling out , freeswitch doesn't get DTMF ? In-Reply-To: <2a18422f.ef8.15c99b7b9bb.Coremail.eastour@163.com> References: <15c898b5d58.2779.b07ebdf329620b8089087c7205b03f01@xbipin.com> <310cf71c.8fc3.15c8ba9c4f6.Coremail.eastour@163.com> <2a18422f.ef8.15c99b7b9bb.Coremail.eastour@163.com> Message-ID: you need to check the DTMF type, you probably are using the wrong one (info-inband-rfc2833), and for some reason they are not negotiated On 12 June 2017 at 02:32, chenyzhi wrote: > I have read the logs ,but I didn't find any difference. > > Please make a test to see if this happens in your box. > > > > > > At 2017-06-10 06:49:34, "David Villasmil" > wrote: > > Have you looked at the log? Bump the logging up and see what shows up... > what you're seeing is very weird > > David > On Fri, Jun 9, 2017 at 11:18 AM chenyzhi wrote: > >> Hi >> >> call 5000 from x-lite , you can hear the IVR voice and if you press dtmf >> keys , freeswitch can receive the dtmf keys . >> >> but , if you enter the command "originate user/1001 5000" , x-lite will >> ring ,answer it ,you can hear the IVR voice , press some keys ,the >> freeswitch can NOT receive any dtmf , >> >> why? >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at 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 at 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 at 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 OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: From vbvbrj at gmail.com Mon Jun 12 12:22:24 2017 From: vbvbrj at gmail.com (Mimiko) Date: Mon, 12 Jun 2017 15:22:24 +0300 Subject: [Freeswitch-users] callcenter member recall same agent. Message-ID: <0f5df8c3-1e59-0a85-82b1-08757a33673c@gmail.com> Hello. I have a question. Is there a parameter that a member, after talking to some agent, leaving callcenter and then call back will be connected specifically to same agent, but be in same queue? If the agent is no more linked to the queue, then member will be connected to any other agent like normally. If agent is still in system and is talking - then member will wait this agent to be free. -- Mimiko desu. From eastour at 163.com Mon Jun 12 13:16:08 2017 From: eastour at 163.com (chenyzhi) Date: Mon, 12 Jun 2017 21:16:08 +0800 (CST) Subject: [Freeswitch-users] When calling out , freeswitch doesn't get DTMF ? In-Reply-To: References: <15c898b5d58.2779.b07ebdf329620b8089087c7205b03f01@xbipin.com> <310cf71c.8fc3.15c8ba9c4f6.Coremail.eastour@163.com> <2a18422f.ef8.15c99b7b9bb.Coremail.eastour@163.com> Message-ID: <6c11dd0f.c2c1.15c9c727dc7.Coremail.eastour@163.com> I don't think It's the DTMF type , because when I dial 5000 from x-lite (which has registered to freeswitch as 1001) ,I can hear the voice and if I press any dtmf on x-lite, freeswitch can recieve the dtmfs. This means that the DTMF type is correct ,otherwise freeswitch coudn't have received the dtmfs; when I enter the command "originate user/1001 5000" at the freeswitch console , my xlite will ring ,and I answered ,I can hear the voice ,I press some dtmf,but freeswitch can NOT receive any dtmf. really weird. At 2017-06-12 17:37:49, "Giovanni Maruzzelli" wrote: you need to check the DTMF type, you probably are using the wrong one (info-inband-rfc2833), and for some reason they are not negotiated On 12 June 2017 at 02:32, chenyzhi wrote: I have read the logs ,but I didn't find any difference. Please make a test to see if this happens in your box. At 2017-06-10 06:49:34, "David Villasmil" wrote: Have you looked at the log? Bump the logging up and see what shows up... what you're seeing is very weird David On Fri, Jun 9, 2017 at 11:18 AM chenyzhi wrote: Hi call 5000 from x-lite , you can hear the IVR voice and if you press dtmf keys , freeswitch can receive the dtmf keys . but , if you enter the command "originate user/1001 5000" , x-lite will ring ,answer it ,you can hear the IVR voice , press some keys ,the freeswitch can NOT receive any dtmf , why? _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at 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 at 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 at 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 at 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 OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: From italo at freeswitch.org Mon Jun 12 15:12:31 2017 From: italo at freeswitch.org (=?UTF-8?B?w410YWxvIFJvc3Np?=) Date: Mon, 12 Jun 2017 12:12:31 -0300 Subject: [Freeswitch-users] callcenter member recall same agent. In-Reply-To: <0f5df8c3-1e59-0a85-82b1-08757a33673c@gmail.com> References: <0f5df8c3-1e59-0a85-82b1-08757a33673c@gmail.com> Message-ID: There's no such feature yet... On Mon, Jun 12, 2017 at 9:22 AM, Mimiko wrote: > Hello. > > I have a question. > > Is there a parameter that a member, after talking to some agent, leaving > callcenter and then call back will be connected specifically to same agent, > but be in same queue? If the agent is no more linked to the queue, then > member will be connected to any other agent like normally. > If agent is still in system and is talking - then member will wait this > agent to be free. > > -- > Mimiko desu. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ítalo Rossi italo at freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From luis.daniel.lucio at gmail.com Mon Jun 12 14:57:17 2017 From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz) Date: Mon, 12 Jun 2017 10:57:17 -0400 Subject: [Freeswitch-users] When calling out , freeswitch doesn't get DTMF ? In-Reply-To: <6c11dd0f.c2c1.15c9c727dc7.Coremail.eastour@163.com> References: <15c898b5d58.2779.b07ebdf329620b8089087c7205b03f01@xbipin.com> <310cf71c.8fc3.15c8ba9c4f6.Coremail.eastour@163.com> <2a18422f.ef8.15c99b7b9bb.Coremail.eastour@163.com> <6c11dd0f.c2c1.15c9c727dc7.Coremail.eastour@163.com> Message-ID: check if you have transcoding, and if you do, check that dftm type-codec on leg b are compatible. -- Luis Daniel Lucio Quiroz CISSP, CISM, CISA Linux, VoIP and much more fun www.okay.com.mx Need LCR? Check out LCR for FusionPBX with FreeSWITCH Need Billing? Check out Billing for FusionPBX with FreeSWITCH On Mon, Jun 12, 2017 at 9:16 AM, chenyzhi wrote: > I don't think It's the DTMF type , because when I dial 5000 from x-lite > (which has registered to freeswitch as 1001) ,I can hear the voice and if I > press any dtmf on x-lite, freeswitch can recieve the dtmfs. This means > that the DTMF type is correct ,otherwise freeswitch coudn't have received > the dtmfs; > > when I enter the command "originate user/1001 5000" at the freeswitch > console , my xlite will ring ,and I answered ,I can hear the voice ,I press > some dtmf,but freeswitch can NOT receive any dtmf. really weird. > > > > > > At 2017-06-12 17:37:49, "Giovanni Maruzzelli" wrote: > > you need to check the DTMF type, you probably are using the wrong one > (info-inband-rfc2833), and for some reason they are not negotiated > > On 12 June 2017 at 02:32, chenyzhi wrote: > >> I have read the logs ,but I didn't find any difference. >> >> Please make a test to see if this happens in your box. >> >> >> >> >> >> At 2017-06-10 06:49:34, "David Villasmil" >> wrote: >> >> Have you looked at the log? Bump the logging up and see what shows up... >> what you're seeing is very weird >> >> David >> On Fri, Jun 9, 2017 at 11:18 AM chenyzhi wrote: >> >>> Hi >>> >>> call 5000 from x-lite , you can hear the IVR voice and if you press dtmf >>> keys , freeswitch can receive the dtmf keys . >>> >>> but , if you enter the command "originate user/1001 5000" , x-lite >>> will ring ,answer it ,you can hear the IVR voice , press some keys ,the >>> freeswitch can NOT receive any dtmf , >>> >>> why? >>> >>> >>> >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at 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 at 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 at 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 at 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 > OpenTelecom.IT > cell: +39 347 266 56 18 > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From igorolhovskiy at gmail.com Mon Jun 12 17:12:36 2017 From: igorolhovskiy at gmail.com (Igor Olhovskiy) Date: Mon, 12 Jun 2017 20:12:36 +0300 Subject: [Freeswitch-users] User CIDR and ACL Message-ID: <75fee20d-577d-41d5-bbdf-77628144691e@Spark> Hi! Can you pls help me with little misunderstanding. According to https://freeswitch.org/confluence/display/FREESWITCH/ACL#ACL-Users I can create ACL for a user and also specify a CIDR in tag. Like So, what is difference between cidr parameter and «auth-acl» parameter, and can I specify, that users can register only from specified network range? Like I want my user 1001 register only from 10.0.20.0/24 and user 1002 register only from 10.0.30.0/24, but they must register with username and pass. Regards, Igor -------------- next part -------------- An HTML attachment was scrubbed... URL: From dig1234 at gmail.com Mon Jun 12 20:32:06 2017 From: dig1234 at gmail.com (Daniel Greenwald) Date: Mon, 12 Jun 2017 16:32:06 -0400 Subject: [Freeswitch-users] RPORT still being sent in TCP calls In-Reply-To: References: Message-ID: We have upgraded to latest STABLE from Ubuntu packages but we are still seeing rport in TCP calls: INVITE sip:xxxxxxx at sip.freeswitch.com;transport=tcp SIP/2.0 Via: SIP/2.0/TCP x.x.x.x;rport;branch=z9hG4bKZDKF5N8QcFZcH Max-Forwards: 70 From: "user" ;tag=Z1pQm54mB22De To: Call-ID: c79656fc-d2d0-4446-90c3-060dabf82fd6 CSeq: 108312748 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.6.10-4-726448d~64bit On Fri, Jun 9, 2017 at 2:56 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > Odd number releases are not stable releases they are dev releases. > The version you quoted is from may 2015 2 years ago. > > The latest stable release is 1.6.17 and latest build release changes every > day. > > > > On Fri, Jun 9, 2017 at 10:50 AM, Daniel Greenwald > wrote: > >> We have noticed that FS is sending RPORT in TCP calls to a gateway. It >> was reported as fixed in this bug: >> https://freeswitch.org/jira/browse/FS-6612 >> >> We are running: >> 1.5.15b+git~20150512T053645Z~9eb887af47~64bit >> >> I am not sure why RPORT is still being sent. Is this there a config >> parameter which needs to be set to suppress the RPORT? Or was this change >> reverted in later versions for some reason. Provider is telling us we >> should not be sending RPORT in TCP... >> >> Any info would be greatly appreciated. >> >> Thanks! >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at 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 ♬ @anthmfs ♬ @FreeSWITCH ♬ > > ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ > http://twitter.com/FreeSWITCH > ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ > * > > ClueCon Weekly Development Call > ☎ sip:888 at conference.freeswitch.org ☎ +19193869900 <(919)%20386-9900> > > https://www.youtube.com/watch?v=9XXgW34t40s > https://www.youtube.com/watch?v=NLaDpGQuZDA > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Mon Jun 12 20:42:42 2017 From: mike at jerris.com (Michael Jerris) Date: Mon, 12 Jun 2017 16:42:42 -0400 Subject: [Freeswitch-users] RPORT still being sent in TCP calls In-Reply-To: References: Message-ID: <8E832DB1-BEF7-4EF1-826F-ED57C966E794@jerris.com> Change is still in there, confirmed. The packages you are using are very old (but should also have that patch in it)… I see that you are supplying modified sip traces. If you can reproduce this on master code, please create a Jira with configuration and full debug logs with sip trace (unmodified) attached. Thanks Mike > On Jun 12, 2017, at 4:32 PM, Daniel Greenwald wrote: > > We have upgraded to latest STABLE from Ubuntu packages but we are still seeing rport in TCP calls: > > > INVITE sip:xxxxxxx at sip.freeswitch.com ;transport=tcp SIP/2.0 > Via: SIP/2.0/TCP x.x.x.x;rport;branch=z9hG4bKZDKF5N8QcFZcH > Max-Forwards: 70 > From: "user" >;tag=Z1pQm54mB22De > To: ;transport=tcp> > Call-ID: c79656fc-d2d0-4446-90c3-060dabf82fd6 > CSeq: 108312748 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.6.10-4-726448d~64bit > > On Fri, Jun 9, 2017 at 2:56 PM, Anthony Minessale > wrote: > Odd number releases are not stable releases they are dev releases. > The version you quoted is from may 2015 2 years ago. > > The latest stable release is 1.6.17 and latest build release changes every day. > > > > On Fri, Jun 9, 2017 at 10:50 AM, Daniel Greenwald > wrote: > We have noticed that FS is sending RPORT in TCP calls to a gateway. It was reported as fixed in this bug: > https://freeswitch.org/jira/browse/FS-6612 > > We are running: > 1.5.15b+git~20150512T053645Z~9eb887af47~64bit > > I am not sure why RPORT is still being sent. Is this there a config parameter which needs to be set to suppress the RPORT? Or was this change reverted in later versions for some reason. Provider is telling us we should not be sending RPORT in TCP... > > Any info would be greatly appreciated. > > Thanks! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at 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 ♬ @anthmfs ♬ @FreeSWITCH ♬ > > ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH > ☞ irc.freenode.net #freeswitch ☞ http://freeswitch.org/g+ > > ClueCon Weekly Development Call > ☎ sip:888 at conference.freeswitch.org ☎ +19193869900 > > https://www.youtube.com/watch?v=9XXgW34t40s > https://www.youtube.com/watch?v=NLaDpGQuZDA > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From prex5609 at gmail.com Tue Jun 13 00:28:44 2017 From: prex5609 at gmail.com (Peter Rex) Date: Mon, 12 Jun 2017 18:28:44 -0600 Subject: [Freeswitch-users] Debian Stretch Message-ID: Stretch is the new stable on Saturday. I've looked through Confluence and the mailing lists but I can't find anything relevant. I see interesting possibilities at http://files.freeswitch.org/repo/deb, but I thought I would ask the mailing list if there's a plan yet to add or move the _production_ build to Stretch. -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Tue Jun 13 00:44:20 2017 From: mike at jerris.com (Michael Jerris) Date: Mon, 12 Jun 2017 20:44:20 -0400 Subject: [Freeswitch-users] Debian Stretch In-Reply-To: References: Message-ID: Stretch won’t build yet. I’ll have some patches over the next few weeks to fix that. 1.8 when released will likely target Stretch as its primary but still a bunch of testing to do. The patches to fix build for stretch will go back into 1.6 branch, once they are complete and tested. > On Jun 12, 2017, at 8:28 PM, Peter Rex wrote: > > Stretch is the new stable on Saturday. I've looked through Confluence and the mailing lists but I can't find anything relevant. I see interesting possibilities at http://files.freeswitch.org/repo/deb , but I thought I would ask the mailing list if there's a plan yet to add or move the _production_ build to Stretch. -------------- next part -------------- An HTML attachment was scrubbed... URL: From prex5609 at gmail.com Tue Jun 13 01:32:31 2017 From: prex5609 at gmail.com (Peter Rex) Date: Mon, 12 Jun 2017 19:32:31 -0600 Subject: [Freeswitch-users] Debian Stretch In-Reply-To: References: Message-ID: Thanks Michael. Hate to do this to you, but is there an estimate on 1.8 timeframe? Mailing list shows people were talking about configs and feature requests in January, but can't see much else. Maybe I'm not looking in the right place. On Mon, Jun 12, 2017 at 6:44 PM, Michael Jerris wrote: > Stretch won’t build yet. I’ll have some patches over the next few weeks > to fix that. 1.8 when released will likely target Stretch as its primary > but still a bunch of testing to do. The patches to fix build for stretch > will go back into 1.6 branch, once they are complete and tested. > > > On Jun 12, 2017, at 8:28 PM, Peter Rex wrote: > > Stretch is the new stable on Saturday. I've looked through Confluence and > the mailing lists but I can't find anything relevant. I see interesting > possibilities at http://files.freeswitch.org/repo/deb, but I thought I > would ask the mailing list if there's a plan yet to add or move the > _production_ build to Stretch. > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Tue Jun 13 02:14:03 2017 From: mike at jerris.com (Michael Jerris) Date: Mon, 12 Jun 2017 22:14:03 -0400 Subject: [Freeswitch-users] Debian Stretch In-Reply-To: References: Message-ID: <733A7BE7-6E66-4BD8-8736-B6DDBFE695A4@jerris.com> announcements will come out when we have real dates. > On Jun 12, 2017, at 9:32 PM, Peter Rex wrote: > > Thanks Michael. Hate to do this to you, but is there an estimate on 1.8 timeframe? Mailing list shows people were talking about configs and feature requests in January, but can't see much else. Maybe I'm not looking in the right place. > > On Mon, Jun 12, 2017 at 6:44 PM, Michael Jerris > wrote: > Stretch won’t build yet. I’ll have some patches over the next few weeks to fix that. 1.8 when released will likely target Stretch as its primary but still a bunch of testing to do. The patches to fix build for stretch will go back into 1.6 branch, once they are complete and tested. > > >> On Jun 12, 2017, at 8:28 PM, Peter Rex > wrote: >> >> Stretch is the new stable on Saturday. I've looked through Confluence and the mailing lists but I can't find anything relevant. I see interesting possibilities at http://files.freeswitch.org/repo/deb , but I thought I would ask the mailing list if there's a plan yet to add or move the _production_ build to Stretch. -------------- next part -------------- An HTML attachment was scrubbed... URL: From vbvbrj at gmail.com Tue Jun 13 06:05:21 2017 From: vbvbrj at gmail.com (Mimiko) Date: Tue, 13 Jun 2017 09:05:21 +0300 Subject: [Freeswitch-users] callcenter member recall same agent. In-Reply-To: References: <0f5df8c3-1e59-0a85-82b1-08757a33673c@gmail.com> Message-ID: <677a8fd6-a4f2-c009-b0cd-52576a5574c1@gmail.com> Ok. There is no such feature. Is it hard to implement if paying? On 12.06.2017 18:12, Ítalo Rossi wrote: > There's no such feature yet... > > On Mon, Jun 12, 2017 at 9:22 AM, Mimiko > wrote: > > Hello. > > I have a question. > > Is there a parameter that a member, after talking to some agent, leaving callcenter and then call back will be connected specifically to same > agent, but be in same queue? If the agent is no more linked to the queue, then member will be connected to any other agent like normally. > If agent is still in system and is talking - then member will wait this agent to be free. -- Mimiko desu. From kbdfck at gmail.com Tue Jun 13 07:52:29 2017 From: kbdfck at gmail.com (Dmitry Sytchev) Date: Tue, 13 Jun 2017 10:52:29 +0300 Subject: [Freeswitch-users] User CIDR and ACL In-Reply-To: <75fee20d-577d-41d5-bbdf-77628144691e@Spark> References: <75fee20d-577d-41d5-bbdf-77628144691e@Spark> Message-ID: With CIDR it is possible to bypass password auth for specified networks 2017-06-12 20:12 GMT+03:00 Igor Olhovskiy : > Hi! > > Can you pls help me with little misunderstanding. > According to > https://freeswitch.org/confluence/display/FREESWITCH/ACL#ACL-Users > I can create ACL for a user and also specify a CIDR in tag. > Like > ;tag=46baee66 To: "sip:500 at freeconf.com";tag=pv4B8Q9XUDtgD Call-ID: 2d118609-1 at 10.1.30.180 CSeq: 1805684444 SUBSCRIBE Max-Forwards: 69 Contact: "RamanTest" User-Agent: TestConference Event: conference Expires: 3600 Allow: INVITE,ACK,BYE,CANCEL,REFER,NOTIFY,OPTIONS,PRACK,UPDATE,INFO,MESSAGE,SUBSCRIBE,PUBLISH Allow-Events: refer, presence Supported: replaces, timer, gruu, join Date: Tue, 13 Jun 2017 08:24:00 GMT Content-Length: 0 ## T 2017/06/13 08:20:14.873026 10.2.30.63:5060 -> 10.1.30.27:55503 [AP] SIP/2.0 202 Accepted Via: SIP/2.0/TCP 10.1.30.199;branch=z9hG4bKc038.62a9c22ab84694b453503a45210a1392.0;i=1;received=10.1.30.27;rport=55503 Via: SIP/2.0/TCP 10.1.30.174;branch=z9hG4bKc038.8d466a0ae4f1821a3f0d08e6602cdadc.0;i=82 Via: SIP/2.0/TLS 10.1.30.146:51890 ;received=10.1.30.146;rport=51890;branch=z9hG4bKd1633fda00007 Record-Route: Record-Route: Record-Route: From: "RamanTest" ;tag=46baee66 To: "sip:500 at freeconf.com" ;tag=pv4B8Q9XUDtgD Call-ID: 2d118609-1 at 10.1.30.180 CSeq: 1805684444 SUBSCRIBE Contact: Expires: 3600 User-Agent: FreeSWITCH-mod_sofia/1.6.17~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=3600 Content-Length: 0 #### T 2017/06/13 08:20:15.173591 10.2.30.63:58879 -> 10.1.30.27:5060 [AP] NOTIFY sip:ramantest at 10.1.30.146:51890;transport=tls SIP/2.0 Via: SIP/2.0/TCP 52.64.221.219;rport;branch=z9hG4bKvr9Kyp8Fe829g Route: ;transport=tcp;ftag=46baee66;lr Record-Route: ;transport=tcp;ftag=46baee66;lr Max-Forwards: 70 From: "sip:500 at freeconf.com" ;tag=pv4B8Q9XUDtgD;tag=pv4B8Q9XUDtgD To: "RamanTest" ;tag=46baee66 Call-ID: 2d118609-1 at 10.1.30.180 CSeq: 705660701 NOTIFY Contact: ;isfocus User-Agent: FreeSWITCH-mod_sofia/1.6.17~64bit Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Event: conference Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=3600 Content-Type: application/conference-info+xml Content-Length: 1028 FreeSWITCH Conference sip:500 at freeconf.com 1 true RamanTest RamanTest connected 2017-06-13T08:20:13+00:00 audio 4048072604 sendrecv -------------- next part -------------- An HTML attachment was scrubbed... URL: From agubbe at gmail.com Tue Jun 13 11:24:26 2017 From: agubbe at gmail.com (=?UTF-8?Q?Agust=C3=AD_Ubalde_Bellot?=) Date: Tue, 13 Jun 2017 13:24:26 +0200 Subject: [Freeswitch-users] Freeswitch sslv3 support Message-ID: Hi all, Is there a FreeSWITCH update where sslv3 support is disabled? Thanks, Agustí -------------- next part -------------- An HTML attachment was scrubbed... URL: From sdpaek21 at gmail.com Tue Jun 13 12:59:31 2017 From: sdpaek21 at gmail.com (Sp Pho) Date: Tue, 13 Jun 2017 15:59:31 +0300 Subject: [Freeswitch-users] B-Leg Early Media - Normal Clearing vs Originator Cancelled Message-ID: Hi, I'm creating a bridge to terminate call arriving inbound from a DID. Everything works fine, however the termination point plays a recorded message - which as rightfully so by the switch is seen as early media (with pre-answer). However, when the originator hangs up while the early media is playing, the CDRs indicate NORMAL_CLEARING. If ignore early media is enabled, the media does not play and ORIGINATOR_CANCELis the hang-up cause. Is there any way to receive the early media and from the CDR know whether or not the call was subsequently answered or not by the agent (i.e. if the originator canceled the call during the early media or while on hold)? Thanks in advance, SP -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian at freeswitch.org Tue Jun 13 13:55:58 2017 From: brian at freeswitch.org (Brian West) Date: Tue, 13 Jun 2017 08:55:58 -0500 Subject: [Freeswitch-users] Freeswitch sslv3 support In-Reply-To: References: Message-ID: You can already disable it via config. Our vanilla config already ships with only 'tlsv1,tlsv1.1,tlsv1.2' enabled. /b On Tue, Jun 13, 2017 at 6:24 AM, Agustí Ubalde Bellot wrote: > Hi all, > > Is there a FreeSWITCH update where sslv3 support is disabled? > > > Thanks, > Agustí > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Book a phone call (CST) Allison prompts for FreeSWITCH: *https://www.gofundme.com/allison-prompts-for-freeswitch* Got Bugs? Report them here ! | Reddit: /r/freeswitch *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian at freeswitch.org Tue Jun 13 13:57:10 2017 From: brian at freeswitch.org (Brian West) Date: Tue, 13 Jun 2017 08:57:10 -0500 Subject: [Freeswitch-users] Record-routes in NOTIFY In-Reply-To: References: Message-ID: Any bug reports belong on JIRA, https://freeswitch.org/confluence/display/FREESWITCH/Reporting+Bugs+to+JIRA Thanks, On Tue, Jun 13, 2017 at 3:35 AM, Ram wrote: > Hi, > > Record routes in SUBSCRIBE is not honored in NOTIFY, In my case i am > having 3 record routes in SUBSCRIBE, but only one i.e top record route is > used for NOTIFY is causing routing issue. I am using freeswitch version > 1.6.17 for testing. > > Following is the trace for SUBSCRIBE and NOTIFY at freeswitch. > > T 2017/06/13 08:20:14.868294 10.1.30.27:55503 -> 10.2.30.63:5060 [AP] > SUBSCRIBE sip:500 at 52.64.221.219:5060;transport=tcp SIP/2.0 > Record-Route: > Record-Route: > Record-Route: > Via: SIP/2.0/TCP 10.1.30.199;branch=z9hG4bKc038. > 62a9c22ab84694b453503a45210a1392.0;i=1 > Via: SIP/2.0/TCP 10.1.30.174;branch=z9hG4bKc038. > 8d466a0ae4f1821a3f0d08e6602cdadc.0;i=82 > Via: SIP/2.0/TLS 10.1.30.146:51890;received=10. > 1.30.146;rport=51890;branch=z9hG4bKd1633fda00007 > From: "RamanTest";tag=46baee66 > To: "sip:500 at freeconf.com" tls>;tag=pv4B8Q9XUDtgD > Call-ID: 2d118609-1 at 10.1.30.180 > CSeq: 1805684444 SUBSCRIBE > Max-Forwards: 69 > Contact: "RamanTest" > User-Agent: TestConference > Event: conference > Expires: 3600 > Allow: INVITE,ACK,BYE,CANCEL,REFER,NOTIFY,OPTIONS,PRACK,UPDATE, > INFO,MESSAGE,SUBSCRIBE,PUBLISH > Allow-Events: refer, presence > Supported: replaces, timer, gruu, join > Date: Tue, 13 Jun 2017 08:24:00 GMT > Content-Length: 0 > > > ## > T 2017/06/13 08:20:14.873026 10.2.30.63:5060 -> 10.1.30.27:55503 [AP] > SIP/2.0 202 Accepted > Via: SIP/2.0/TCP 10.1.30.199;branch=z9hG4bKc038. > 62a9c22ab84694b453503a45210a1392.0;i=1;received=10.1.30.27;rport=55503 > Via: SIP/2.0/TCP 10.1.30.174;branch=z9hG4bKc038. > 8d466a0ae4f1821a3f0d08e6602cdadc.0;i=82 > Via: SIP/2.0/TLS 10.1.30.146:51890;received=10. > 1.30.146;rport=51890;branch=z9hG4bKd1633fda00007 > Record-Route: > Record-Route: > Record-Route: > From: "RamanTest" ;tag=46baee66 > To: "sip:500 at freeconf.com" ;tag= > pv4B8Q9XUDtgD > Call-ID: 2d118609-1 at 10.1.30.180 > CSeq: 1805684444 SUBSCRIBE > Contact: > Expires: 3600 > User-Agent: FreeSWITCH-mod_sofia/1.6.17~64bit > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, > REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, path, replaces > Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, > line-seize, call-info, sla, include-session-description, presence.winfo, > message-summary, refer > Subscription-State: active;expires=3600 > Content-Length: 0 > > > #### > T 2017/06/13 08:20:15.173591 10.2.30.63:58879 -> 10.1.30.27:5060 [AP] > NOTIFY sip:ramantest at 10.1.30.146:51890;transport=tls SIP/2.0 > Via: SIP/2.0/TCP 52.64.221.219;rport;branch=z9hG4bKvr9Kyp8Fe829g > Route: ;transport=tcp;ftag=46baee66;lr > Record-Route: ;transport=tcp;ftag=46baee66;lr > Max-Forwards: 70 > From: "sip:500 at freeconf.com" ;tag= > pv4B8Q9XUDtgD;tag=pv4B8Q9XUDtgD > To: "RamanTest" ;tag=46baee66 > Call-ID: 2d118609-1 at 10.1.30.180 > CSeq: 705660701 NOTIFY > Contact: ;isfocus > User-Agent: FreeSWITCH-mod_sofia/1.6.17~64bit > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, > REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, path, replaces > Event: conference > Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, > line-seize, call-info, sla, include-session-description, presence.winfo, > message-summary, refer > Subscription-State: active;expires=3600 > Content-Type: application/conference-info+xml > Content-Length: 1028 > > > entity="sip:500 at freeconf.com"> > > FreeSWITCH Conference > > > sip:500 at freeconf.com > > > > > 1 > true > > > > RamanTest > > RamanTest > connected > > 2017-06-13T08:20:13+00:00 > > > audio > 4048072604 <(404)%20807-2604> > sendrecv > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Book a phone call (CST) Allison prompts for FreeSWITCH: *https://www.gofundme.com/allison-prompts-for-freeswitch* Got Bugs? Report them here ! | Reddit: /r/freeswitch *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: From joel at gogii.net Tue Jun 13 14:38:57 2017 From: joel at gogii.net (Joel Serrano) Date: Tue, 13 Jun 2017 07:38:57 -0700 Subject: [Freeswitch-users] Multiple FreeSWITCH servers behind kamailio-websocket In-Reply-To: References: Message-ID: Hi Karsten, Have you tried with regular Kamailio (w/ dispatcher+websocket+xhttp modules)? I don't see why it wouldn't work... Joel. On Wed, Jun 7, 2017 at 4:47 AM, Karsten Horsmann wrote: > Hello List, > > > is there any howto about webrtc loadbalance in combination with kamailio > and FreeSWITCH? > > I want to share one WSS address/endpoint to multiple FreeSWITCH backends. > Or is there any other best practice? > > My callflow is mostly that my internal SIP Servers called my registered > webrtc clients. > > Would be nice to get some input. > > -- > Kind Regards > *Karsten Horsmann* > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From eastour at 163.com Tue Jun 13 14:19:07 2017 From: eastour at 163.com (chenyzhi) Date: Tue, 13 Jun 2017 22:19:07 +0800 (CST) Subject: [Freeswitch-users] When calling out , freeswitch doesn't get DTMF ? In-Reply-To: References: <15c898b5d58.2779.b07ebdf329620b8089087c7205b03f01@xbipin.com> <310cf71c.8fc3.15c8ba9c4f6.Coremail.eastour@163.com> <2a18422f.ef8.15c99b7b9bb.Coremail.eastour@163.com> <6c11dd0f.c2c1.15c9c727dc7.Coremail.eastour@163.com> Message-ID: <2f48505d.d9c5.15ca1d28424.Coremail.eastour@163.com> It's a one leg call .there is no b-leg. please make a test on your freeswitch box . just type the command "originate user/1001 5000" on the freeswitch console to see if your freeswitch instance can detect dtmf input. At 2017-06-12 22:57:17, "Luis Daniel Lucio Quiroz" wrote: check if you have transcoding, and if you do, check that dftm type-codec on leg b are compatible. -- Luis Daniel Lucio Quiroz CISSP, CISM, CISA Linux, VoIP and much more fun www.okay.com.mx Need LCR? Check out LCR for FusionPBX with FreeSWITCH Need Billing? Check out Billing for FusionPBX with FreeSWITCH On Mon, Jun 12, 2017 at 9:16 AM, chenyzhi wrote: I don't think It's the DTMF type , because when I dial 5000 from x-lite (which has registered to freeswitch as 1001) ,I can hear the voice and if I press any dtmf on x-lite, freeswitch can recieve the dtmfs. This means that the DTMF type is correct ,otherwise freeswitch coudn't have received the dtmfs; when I enter the command "originate user/1001 5000" at the freeswitch console , my xlite will ring ,and I answered ,I can hear the voice ,I press some dtmf,but freeswitch can NOT receive any dtmf. really weird. At 2017-06-12 17:37:49, "Giovanni Maruzzelli" wrote: you need to check the DTMF type, you probably are using the wrong one (info-inband-rfc2833), and for some reason they are not negotiated On 12 June 2017 at 02:32, chenyzhi wrote: I have read the logs ,but I didn't find any difference. Please make a test to see if this happens in your box. At 2017-06-10 06:49:34, "David Villasmil" wrote: Have you looked at the log? Bump the logging up and see what shows up... what you're seeing is very weird David On Fri, Jun 9, 2017 at 11:18 AM chenyzhi wrote: Hi call 5000 from x-lite , you can hear the IVR voice and if you press dtmf keys , freeswitch can receive the dtmf keys . but , if you enter the command "originate user/1001 5000" , x-lite will ring ,answer it ,you can hear the IVR voice , press some keys ,the freeswitch can NOT receive any dtmf , why? _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at 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 at 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 at 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 at 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 OpenTelecom.IT cell: +39 347 266 56 18 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From zzyroy at qq.com Tue Jun 13 14:35:18 2017 From: zzyroy at qq.com (=?ISO-8859-1?B?enp5?=) Date: Tue, 13 Jun 2017 22:35:18 +0800 Subject: [Freeswitch-users] How to use originate make A_leg to a queue Message-ID: Dear All, I'm testing mod_callcenter now. Is there any way to make A_leg to the call center queue first then bridge B_leg by dialplan? Just like [ originate group/sales+A 8888 XML default ] Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Tue Jun 13 19:32:24 2017 From: david.villasmil.work at gmail.com (David Villasmil) Date: Tue, 13 Jun 2017 19:32:24 +0000 Subject: [Freeswitch-users] How to use originate make A_leg to a queue In-Reply-To: References: Message-ID: Originate blahnlah &callcenter(yourqueue) Or something like that On Tue, Jun 13, 2017 at 3:30 PM zzy wrote: > Dear All, > > I'm testing mod_callcenter now. > > Is there any way to make A_leg to the call center queue first then bridge > B_leg by dialplan? > > Just like [ originate group/sales+A 8888 XML default ] > > Thanks. > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From admin at tedssupply.com Tue Jun 13 21:06:26 2017 From: admin at tedssupply.com (admin) Date: Tue, 13 Jun 2017 17:06:26 -0400 Subject: [Freeswitch-users] When calling out , freeswitch doesn't get DTMF ? In-Reply-To: <2f48505d.d9c5.15ca1d28424.Coremail.eastour@163.com> References: <15c898b5d58.2779.b07ebdf329620b8089087c7205b03f01@xbipin.com> <310cf71c.8fc3.15c8ba9c4f6.Coremail.eastour@163.com> <2a18422f.ef8.15c99b7b9bb.Coremail.eastour@163.com> <6c11dd0f.c2c1.15c9c727dc7.Coremail.eastour@163.com> <2f48505d.d9c5.15ca1d28424.Coremail.eastour@163.com> Message-ID: <59401B92020000310000A34D@mail.tedssupply.com> I am encountering the same issue. I am using an ESL call to api originate, with little else except getting in and out of the port, and the caller DTMF is failing to reach the called number. The called number auto answer attendant does not respond to DTMF, and a call to a test phone confirms no DTMF. This is a new upgrade from 1.2 to 1.6 and I don't know if this was an issue in 1.2, but my users never complained before 1.6 upgrade. Ideas?... >>> chenyzhi 06/13/17 3:32 PM >>> It's a one leg call .there is no b-leg. please make a test on your freeswitch box . just type the command "originate user/1001 5000" on the freeswitch console to see if your freeswitch instance can detect dtmf input. At 2017-06-12 22:57:17, "Luis Daniel Lucio Quiroz" wrote: check if you have transcoding, and if you do, check that dftm type-codec on leg b are compatible. -- Luis Daniel Lucio Quiroz CISSP, CISM, CISA Linux, VoIP and much more fun www.okay.com.mx Need LCR? Check out LCR for FusionPBX with FreeSWITCH Need Billing? Check out Billing for FusionPBX with FreeSWITCH On Mon, Jun 12, 2017 at 9:16 AM, chenyzhi wrote: I don't think It's the DTMF type , because when I dial 5000 from x-lite (which has registered to freeswitch as 1001) ,I can hear the voice and if I press any dtmf on x-lite, freeswitch can recieve the dtmfs. This means that the DTMF type is correct ,otherwise freeswitch coudn't have received the dtmfs; when I enter the command "originate user/1001 5000" at the freeswitch console , my xlite will ring ,and I answered ,I can hear the voice ,I press some dtmf,but freeswitch can NOT receive any dtmf. really weird. At 2017-06-12 17:37:49, "Giovanni Maruzzelli" wrote: you need to check the DTMF type, you probably are using the wrong one (info-inband-rfc2833), and for some reason they are not negotiated On 12 June 2017 at 02:32, chenyzhi wrote: I have read the logs ,but I didn't find any difference. Please make a test to see if this happens in your box. At 2017-06-10 06:49:34, "David Villasmil" wrote: Have you looked at the log? Bump the logging up and see what shows up... what you're seeing is very weird David On Fri, Jun 9, 2017 at 11:18 AM chenyzhi wrote: Hi call 5000 from x-lite , you can hear the IVR voice and if you press dtmf keys , freeswitch can receive the dtmf keys . but , if you enter the command "originate user/1001 5000" , x-lite will ring ,answer it ,you can hear the IVR voice , press some keys ,the freeswitch can NOT receive any dtmf , why? _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at 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 at 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 at 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 at 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 OpenTelecom.IT cell: +39 347 266 56 18 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From gmaruzz at gmail.com Wed Jun 14 07:53:17 2017 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Wed, 14 Jun 2017 09:53:17 +0200 Subject: [Freeswitch-users] When calling out , freeswitch doesn't get DTMF ? In-Reply-To: <59401B92020000310000A34D@mail.tedssupply.com> References: <15c898b5d58.2779.b07ebdf329620b8089087c7205b03f01@xbipin.com> <310cf71c.8fc3.15c8ba9c4f6.Coremail.eastour@163.com> <2a18422f.ef8.15c99b7b9bb.Coremail.eastour@163.com> <6c11dd0f.c2c1.15c9c727dc7.Coremail.eastour@163.com> <2f48505d.d9c5.15ca1d28424.Coremail.eastour@163.com> <59401B92020000310000A34D@mail.tedssupply.com> Message-ID: Never heard such problems Please pastebin your dialplan, your SIP profile, and the complete, since beginning to end, unedited, debug output of console when receiving a call which does not get DTMFs sent from mobile cell: +39 347 266 56 18 Giovanni Maruzzelli OpenTelecom.IT On Jun 13, 2017 23:07, "admin" wrote: > I am encountering the same issue. I am using an ESL call to api > originate, with little else except getting in and out of the port, and the > caller DTMF is failing to reach the called number. The called number auto > answer attendant does not respond to DTMF, and a call to a test phone > confirms no DTMF. This is a new upgrade from 1.2 to 1.6 and I don't know > if this was an issue in 1.2, but my users never complained before 1.6 > upgrade. Ideas?... > > > >>> chenyzhi 06/13/17 3:32 PM >>> > It's a one leg call .there is no b-leg. > > please make a test on your freeswitch box . > > just type the command "originate user/1001 5000" on the freeswitch console > to see if your freeswitch instance can detect dtmf input. > > > > > > At 2017-06-12 22:57:17, "Luis Daniel Lucio Quiroz" < > luis.daniel.lucio at gmail.com> wrote: > > check if you have transcoding, and if you do, check that dftm type-codec > on leg b are compatible. > > -- > Luis Daniel Lucio Quiroz > CISSP, CISM, CISA > Linux, VoIP and much more fun > www.okay.com.mx > > Need LCR? Check out LCR for FusionPBX with FreeSWITCH > Need Billing? Check out Billing for FusionPBX with FreeSWITCH > > On Mon, Jun 12, 2017 at 9:16 AM, chenyzhi wrote: > >> I don't think It's the DTMF type , because when I dial 5000 from x-lite >> (which has registered to freeswitch as 1001) ,I can hear the voice and if I >> press any dtmf on x-lite, freeswitch can recieve the dtmfs. This means >> that the DTMF type is correct ,otherwise freeswitch coudn't have received >> the dtmfs; >> >> when I enter the command "originate user/1001 5000" at the freeswitch >> console , my xlite will ring ,and I answered ,I can hear the voice ,I press >> some dtmf,but freeswitch can NOT receive any dtmf. really weird. >> >> >> >> >> >> At 2017-06-12 17:37:49, "Giovanni Maruzzelli" wrote: >> >> you need to check the DTMF type, you probably are using the wrong one >> (info-inband-rfc2833), and for some reason they are not negotiated >> >> On 12 June 2017 at 02:32, chenyzhi wrote: >> >>> I have read the logs ,but I didn't find any difference. >>> >>> Please make a test to see if this happens in your box. >>> >>> >>> >>> >>> >>> At 2017-06-10 06:49:34, "David Villasmil" >> com> wrote: >>> >>> Have you looked at the log? Bump the logging up and see what shows up... >>> what you're seeing is very weird >>> >>> David >>> On Fri, Jun 9, 2017 at 11:18 AM chenyzhi wrote: >>> >>>> Hi >>>> >>>> call 5000 from x-lite , you can hear the IVR voice and if you press >>>> dtmf keys , freeswitch can receive the dtmf keys . >>>> >>>> but , if you enter the command "originate user/1001 5000" , x-lite >>>> will ring ,answer it ,you can hear the IVR voice , press some keys ,the >>>> freeswitch can NOT receive any dtmf , >>>> >>>> why? >>>> >>>> >>>> >>>> >>>> ____________________________________________________________ >>>> _____________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at 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 at 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 at 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 at 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 >> OpenTelecom.IT >> cell: +39 347 266 56 18 >> >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From miconda at gmail.com Wed Jun 14 08:57:00 2017 From: miconda at gmail.com (Daniel-Constantin Mierla) Date: Wed, 14 Jun 2017 10:57:00 +0200 Subject: [Freeswitch-users] [SR-Users] Multiple FreeSWITCH servers behind kamailio-websocket In-Reply-To: References: Message-ID: <407012cb-dc25-b237-17d9-69462307e05b@gmail.com> Hello, that combination of dispatcher+websocket+xhttp modules works just fine... So to load balance the SIP signaling with Kamailio towards FreeSwitch, just use the dispatcher module as usual. A sample config is available at: - https://www.kamailio.org/docs/modules/stable/modules/dispatcher.html#dispatcher.ex.config You need to add the support for websocket traffic via websocket module: - https://www.kamailio.org/docs/modules/stable/modules/websocket.html#idp42826164 or extract from the tutorial linked in a previous email on this thread. Cheers, Daniel On 13.06.17 16:38, Joel Serrano wrote: > Hi Karsten, > > Have you tried with regular Kamailio (w/ dispatcher+websocket+xhttp > modules)? I don't see why it wouldn't work... > > Joel. > > > On Wed, Jun 7, 2017 at 4:47 AM, Karsten Horsmann > wrote: > > Hello List, > > > is there any howto about webrtc loadbalance in combination with > kamailio and FreeSWITCH? > > I want to share one WSS address/endpoint to multiple FreeSWITCH > backends. > Or is there any other best practice? > > My callflow is mostly that my internal SIP Servers called my > registered webrtc clients. > > Would be nice to get some input. > > -- > Kind Regards > *Karsten Horsmann* > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > _______________________________________________ > Kamailio (SER) - Users Mailing List > sr-users at lists.kamailio.org > https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla www.twitter.com/miconda -- www.linkedin.com/in/miconda Kamailio Advanced Training - www.asipto.com Kamailio World Conference - www.kamailioworld.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From nandy1925 at gmail.com Wed Jun 14 09:43:30 2017 From: nandy1925 at gmail.com (Nandy Dagondon) Date: Wed, 14 Jun 2017 09:43:30 +0000 Subject: [Freeswitch-users] Paging In-Reply-To: References: <8AE4D7F5-1AE6-44F8-B85D-13C4FADC22CA@magicmail.mooo.com> Message-ID: I've done this using mod_portaudio with auto-answer from this example: https://wiki.freeswitch.org/wiki/Mod_portaudio#PA_System_w.2F_Chime /nandy On Fri, May 19, 2017 at 12:04 PM, Lesley Pervis wrote: > I've done this in Lua with outbound conference calls, but it was pretty > complicated and you have to have endpoints that will auto-answer. > https://freeswitch.org/confluence/display/FREESWITCH/ > Outbound+Conference+Calls > > On Wed, May 17, 2017 at 10:02 AM, Rick Jarvis > wrote: > >> Looking to set up paging (not multicast). What’s the best way of >> achieving this? Specifically, I want to have the receiving handset(s) >> answer muted for privacy reasons, so it’s literally like a PA system rather >> than just auto answer…? >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From rick at magicmail.mooo.com Wed Jun 14 11:03:00 2017 From: rick at magicmail.mooo.com (Rick Jarvis) Date: Wed, 14 Jun 2017 12:03:00 +0100 Subject: [Freeswitch-users] Paging In-Reply-To: References: <8AE4D7F5-1AE6-44F8-B85D-13C4FADC22CA@magicmail.mooo.com> Message-ID: <5D4CCF18-DAA0-4F84-8203-4F69D66B251A@magicmail.mooo.com> Thanks Nandy. We’ve used PA for stuff, but what we’re trying to accomplish here is paging through handsets. It’s kind of strange that there isn’t more support for this as it’s quite a standard feature even on the old analog PBX systems, Panasonic and so on. > On 14 Jun 2017, at 10:43, Nandy Dagondon wrote: > > I've done this using mod_portaudio with auto-answer from this example: > https://wiki.freeswitch.org/wiki/Mod_portaudio#PA_System_w.2F_Chime > > /nandy > > On Fri, May 19, 2017 at 12:04 PM, Lesley Pervis > wrote: > I've done this in Lua with outbound conference calls, but it was pretty complicated and you have to have endpoints that will auto-answer. https://freeswitch.org/confluence/display/FREESWITCH/Outbound+Conference+Calls > > On Wed, May 17, 2017 at 10:02 AM, Rick Jarvis > wrote: > Looking to set up paging (not multicast). What’s the best way of achieving this? Specifically, I want to have the receiving handset(s) answer muted for privacy reasons, so it’s literally like a PA system rather than just auto answer…? > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From shaun.stokes at itec-support.co.uk Wed Jun 14 11:11:00 2017 From: shaun.stokes at itec-support.co.uk (Shaun Stokes) Date: Wed, 14 Jun 2017 11:11:00 +0000 Subject: [Freeswitch-users] Add long contact URI support for mod_callcenter Message-ID: <6FD2F8B5BB72834E9939AEDF9FB802A901E8671F59@mbx-01.sysconfig.co.uk> Currently we're unable to exceed the 255 character limit on the contact field for mod_callcenter which is required to support long contact URIs for agents in mod_callcenter. FreeSWITCH and mod_callcenter fully support long contact URIs (up to 510 characters) as tested in our production environment over the last 6 months. We've submitted a pull request to resolve this problem: https://freeswitch.org/stash/projects/FS/repos/freeswitch/pull-requests/1165/diff Any chance this can be committed? [http://www.itec-support.co.uk/wp-content/uploads/2016/07/email_logo.jpg] Shaun Stokes - Infrastructure Analyst T : 01453 700713 E : shaun.stokes at itec-support.co.uk W : www.itec-support.co.uk Registered Address :- ITEC Support, Suite 2 Prospect House, Bath Road, Stroud, Gloucestershire GL5 3QF Company No. 06908001 CONFIDENTIALITY NOTICE This communication and the information it contains are intended for the person or organisation to which it is addressed. Its contents are confidential and may be protected in law. Unauthorised use, copying or disclosure of any of it may be unlawful. If you are not the intended recipient, please contact us immediately. The contents of any attachments in this e-mail may contain software viruses, which could damage your own computer system. While ITEC Support has taken every reasonable precaution to minimise this risk, we cannot accept liability for any damage which you sustain as a result of software viruses. You should carry out your own virus checking procedure before opening any attachment. ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ From hardikitpl at gmail.com Wed Jun 14 11:52:23 2017 From: hardikitpl at gmail.com (Hardik Patel) Date: Wed, 14 Jun 2017 17:22:23 +0530 Subject: [Freeswitch-users] Fax receive issue with t30 codec In-Reply-To: References: Message-ID: Hi Brian, Is there any solution for that? On Sat, Jun 10, 2017 at 9:46 AM, Hardik Patel wrote: > Hi Brian, > > Thanks for the support. > > We are testing receive fax functionality using real fax machine and here i > have listed the model which we are using to send fax. > > the models of real fax machines that we have used are group 3 CCITT / ITU, > they are the following: > 1 ) konica minolta bizhub-c220 > 2 ) HP Officejet 4500 > 3 ) HP Officejet G85 > > > From above list one of our machine is sending fax without T38 support and > we got the failure with same error which have posted on bug but if we use > T38 support then it works fine for us. > > > > *ERROR :* > 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:508 Fax processing > not successful - result (3) Timed out waiting for the first message. > 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:513 Remote station > id: > 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:514 Local station > id: SpanDSP Fax Ident > 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:515 Pages > transferred: 0 > 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:517 Total fax > pages: 0 > 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:518 Image > resolution: 0x0 > 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:519 Transfer > Rate: 14400 > 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:521 ECM > status off > 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:522 remote country: > 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:523 remote vendor: > 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:524 remote model: > 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:526 > ============================== > > On Sat, Jun 10, 2017 at 2:42 AM, Brian West wrote: > >> Your log snip doesn't really help, I know without a single doubt faxing >> works fine. So what are you doing and how are you doing it? >> >> On Fri, Jun 9, 2017 at 3:42 PM, Tihomir Culjaga >> wrote: >> >>> hi, whats new on faxing in 1.6.17 ? >>> >>> T. >>> >>> On 6 June 2017 at 16:00, Brian West wrote: >>> >>>> You'll have to use 1.6.17 if you ever want any faxing to work in all >>>> test cases. >>>> >>>> /b >>>> >>>> On Tue, Jun 6, 2017 at 6:11 AM, Hardik Patel >>>> wrote: >>>> >>>>> Hello, >>>>> >>>>> I am using opensips as entry point using dispatcher. opensips( >>>>> 127.0.0.1), i am routing call to freeswitch server (127.0.0.3). >>>>> >>>>> Now I am trying to receive fax, my issue is when i try to send fax in >>>>> softphone(Zoiper) from the log i am seeing that it is sending fax using t30 >>>>> codec. and i am not receiving the fax at destination, is it because of >>>>> codec, should it only work with t38 codec? if that is the issue than how am >>>>> i be able to send the fax using t38 from zoiper? >>>>> >>>>> Here i am attaching the fs log with loglevel 9 and sip trace is also >>>>> enabled. >>>>> >>>>> 127.0.0.2 => carrier/provider IP >>>>> 123456789 => Fax number >>>>> test at gamil.com => Email Address >>>>> 127.0.0.4 =>UI IP >>>>> >>>>> Pastebin link:https://pastebin.freeswitch.org/view/9ec52715 >>>>> >>>>> ____________________________________________________________ >>>>> _____________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at 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 at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>>> switch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> >>>> -- >>>> >>>> *Brian West* >>>> brian at freeswitch.org >>>> >>>> *Twitter: @FreeSWITCH , @briankwest* >>>> >>>> http://www.freeswitchbook.com >>>> http://www.freeswitchcookbook.com >>>> >>>> Book a phone call (CST) >>>> >>>> Allison prompts for FreeSWITCH: >>>> >>>> *https://www.gofundme.com/allison-prompts-for-freeswitch* >>>> >>>> >>>> Got Bugs? Report them here ! | Reddit: >>>> /r/freeswitch >>>> >>>> *T:*+19184209001 <+1%20918-420-9001> | *F:*+19184209002 >>>> <+1%20918-420-9002> | *M:*+1918424WEST (9378) >>>> *Skype:*briankwest >>>> >>>> ____________________________________________________________ >>>> _____________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at 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 at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>> switch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at 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 at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> >> *Brian West* >> brian at freeswitch.org >> >> *Twitter: @FreeSWITCH , @briankwest* >> >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> Book a phone call (CST) >> >> Allison prompts for FreeSWITCH: >> >> *https://www.gofundme.com/allison-prompts-for-freeswitch* >> >> >> Got Bugs? Report them here ! | Reddit: >> /r/freeswitch >> >> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) >> *Skype:*briankwest >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Hardik Patel > iNextrix Technologies Pvt Ltd > -- Hardik Patel iNextrix Technologies Pvt Ltd -------------- next part -------------- An HTML attachment was scrubbed... URL: From gmaruzz at gmail.com Wed Jun 14 11:58:25 2017 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Wed, 14 Jun 2017 13:58:25 +0200 Subject: [Freeswitch-users] Fax receive issue with t30 codec In-Reply-To: References: Message-ID: try setting the transfer to 9600 (fax_disable_v17) Also, you may find this page useful: https://freeswitch.org/confluence/display/FREESWITCH/mod_spandsp#mod_spandsp-Fax On 14 June 2017 at 13:52, Hardik Patel wrote: > Hi Brian, > > Is there any solution for that? > > On Sat, Jun 10, 2017 at 9:46 AM, Hardik Patel > wrote: > >> Hi Brian, >> >> Thanks for the support. >> >> We are testing receive fax functionality using real fax machine and here >> i have listed the model which we are using to send fax. >> >> the models of real fax machines that we have used are group 3 CCITT / >> ITU, they are the following: >> 1 ) konica minolta bizhub-c220 >> 2 ) HP Officejet 4500 >> 3 ) HP Officejet G85 >> >> >> From above list one of our machine is sending fax without T38 support and >> we got the failure with same error which have posted on bug but if we use >> T38 support then it works fine for us. >> >> >> >> *ERROR :* >> 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:508 Fax processing >> not successful - result (3) Timed out waiting for the first message. >> 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:513 Remote station >> id: >> 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:514 Local station >> id: SpanDSP Fax Ident >> 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:515 Pages >> transferred: 0 >> 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:517 Total fax >> pages: 0 >> 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:518 Image >> resolution: 0x0 >> 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:519 Transfer >> Rate: 14400 >> 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:521 ECM >> status off >> 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:522 remote >> country: >> 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:523 remote >> vendor: >> 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:524 remote >> model: >> 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:526 >> ============================== >> >> On Sat, Jun 10, 2017 at 2:42 AM, Brian West wrote: >> >>> Your log snip doesn't really help, I know without a single doubt faxing >>> works fine. So what are you doing and how are you doing it? >>> >>> On Fri, Jun 9, 2017 at 3:42 PM, Tihomir Culjaga >>> wrote: >>> >>>> hi, whats new on faxing in 1.6.17 ? >>>> >>>> T. >>>> >>>> On 6 June 2017 at 16:00, Brian West wrote: >>>> >>>>> You'll have to use 1.6.17 if you ever want any faxing to work in all >>>>> test cases. >>>>> >>>>> /b >>>>> >>>>> On Tue, Jun 6, 2017 at 6:11 AM, Hardik Patel >>>>> wrote: >>>>> >>>>>> Hello, >>>>>> >>>>>> I am using opensips as entry point using dispatcher. opensips( >>>>>> 127.0.0.1), i am routing call to freeswitch server (127.0.0.3). >>>>>> >>>>>> Now I am trying to receive fax, my issue is when i try to send fax in >>>>>> softphone(Zoiper) from the log i am seeing that it is sending fax using t30 >>>>>> codec. and i am not receiving the fax at destination, is it because of >>>>>> codec, should it only work with t38 codec? if that is the issue than how am >>>>>> i be able to send the fax using t38 from zoiper? >>>>>> >>>>>> Here i am attaching the fs log with loglevel 9 and sip trace is also >>>>>> enabled. >>>>>> >>>>>> 127.0.0.2 => carrier/provider IP >>>>>> 123456789 => Fax number >>>>>> test at gamil.com => Email Address >>>>>> 127.0.0.4 =>UI IP >>>>>> >>>>>> Pastebin link:https://pastebin.freeswitch.org/view/9ec52715 >>>>>> >>>>>> ____________________________________________________________ >>>>>> _____________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at 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 at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>>>> switch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>> >>>>> >>>>> >>>>> -- >>>>> >>>>> *Brian West* >>>>> brian at freeswitch.org >>>>> >>>>> *Twitter: @FreeSWITCH , @briankwest* >>>>> >>>>> http://www.freeswitchbook.com >>>>> http://www.freeswitchcookbook.com >>>>> >>>>> Book a phone call (CST) >>>>> >>>>> Allison prompts for FreeSWITCH: >>>>> >>>>> *https://www.gofundme.com/allison-prompts-for-freeswitch* >>>>> >>>>> >>>>> Got Bugs? Report them here ! | Reddit: >>>>> /r/freeswitch >>>>> >>>>> *T:*+19184209001 <+1%20918-420-9001> | *F:*+19184209002 >>>>> <+1%20918-420-9002> | *M:*+1918424WEST (9378) >>>>> *Skype:*briankwest >>>>> >>>>> ____________________________________________________________ >>>>> _____________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at 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 at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>>> switch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> ____________________________________________________________ >>>> _____________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at 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 at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>> switch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> >>> -- >>> >>> *Brian West* >>> brian at freeswitch.org >>> >>> *Twitter: @FreeSWITCH , @briankwest* >>> >>> http://www.freeswitchbook.com >>> http://www.freeswitchcookbook.com >>> >>> Book a phone call (CST) >>> >>> Allison prompts for FreeSWITCH: >>> >>> *https://www.gofundme.com/allison-prompts-for-freeswitch* >>> >>> >>> Got Bugs? Report them here ! | Reddit: >>> /r/freeswitch >>> >>> *T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002 >>> <(918)%20420-9002> | *M:*+1918424WEST (9378) >>> *Skype:*briankwest >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at 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 at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Hardik Patel >> iNextrix Technologies Pvt Ltd >> > > > > -- > Hardik Patel > iNextrix Technologies Pvt Ltd > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at 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 OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: From italo at freeswitch.org Wed Jun 14 12:11:48 2017 From: italo at freeswitch.org (=?UTF-8?B?w410YWxvIFJvc3Np?=) Date: Wed, 14 Jun 2017 09:11:48 -0300 Subject: [Freeswitch-users] Add long contact URI support for mod_callcenter In-Reply-To: <6FD2F8B5BB72834E9939AEDF9FB802A901E8671F59@mbx-01.sysconfig.co.uk> References: <6FD2F8B5BB72834E9939AEDF9FB802A901E8671F59@mbx-01.sysconfig.co.uk> Message-ID: Sure, I'll merge it later today. Thank you. On Wed, Jun 14, 2017 at 8:11 AM, Shaun Stokes < shaun.stokes at itec-support.co.uk> wrote: > Currently we're unable to exceed the 255 character limit on the contact > field for mod_callcenter which is required to support long contact URIs for > agents in mod_callcenter. > > FreeSWITCH and mod_callcenter fully support long contact URIs (up to 510 > characters) as tested in our production environment over the last 6 months. > > We've submitted a pull request to resolve this problem: > https://freeswitch.org/stash/projects/FS/repos/freeswitch/ > pull-requests/1165/diff > > Any chance this can be committed? > [http://www.itec-support.co.uk/wp-content/uploads/2016/07/email_logo.jpg] > Shaun Stokes - Infrastructure Analyst > > T : 01453 700713 > E : shaun.stokes at itec-support.co.uk > W : www.itec-support.co.uk > > Registered Address :- ITEC Support, Suite 2 Prospect House, Bath Road, > Stroud, Gloucestershire GL5 3QF > Company No. 06908001 > > CONFIDENTIALITY NOTICE > This communication and the information it contains are intended for the > person or organisation to which it is addressed. Its contents are > confidential and may be protected in law. Unauthorised use, copying or > disclosure of any of it may be unlawful. If you are not the intended > recipient, please contact us immediately. > The contents of any attachments in this e-mail may contain software > viruses, which could damage your own computer system. While ITEC Support > has taken every reasonable precaution to minimise this risk, we cannot > accept liability for any damage which you sustain as a result of software > viruses. You should carry out your own virus checking procedure before > opening any attachment. > > ______________________________________________________________________ > This message has been checked for all known viruses by MessageLabs Virus > Scanning Service. > ______________________________________________________________________ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ítalo Rossi italo at freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From zzyroy at qq.com Wed Jun 14 00:57:40 2017 From: zzyroy at qq.com (=?gb18030?B?enp5?=) Date: Wed, 14 Jun 2017 08:57:40 +0800 Subject: [Freeswitch-users] How to use originate make A_leg to a queue Message-ID: Dear David, Thank you for your reply. This way is put the call center queue in the B leg. But I want to call the queue first (put the queue in the A leg). ------------------ Original ------------------ From: David Villasmil Date: 周三,6月 14,2017 03:36 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to use originate make A_leg to a queue Originate blahnlah &callcenter(yourqueue) Or something like that On Tue, Jun 13, 2017 at 3:30 PM zzy wrote: Dear All, I'm testing mod_callcenter now. Is there any way to make A_leg to the call center queue first then bridge B_leg by dialplan? Just like [ originate group/sales+A 8888 XML default ] Thanks. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From sdpaek21 at gmail.com Wed Jun 14 10:30:54 2017 From: sdpaek21 at gmail.com (Sp Pho) Date: Wed, 14 Jun 2017 13:30:54 +0300 Subject: [Freeswitch-users] B-Leg Early Media - Normal Clearing vs Originator Cancelled In-Reply-To: References: Message-ID: I've pulled a log of the bad case vs good one. A diff of both can be found here: https://www.diffchecker.com/zWNNGibE. This is from the same originating number - any ideas why on line 96, early media is detected for the good case and not the bad, would be greatly appreciated. thanks in advance On Tue, Jun 13, 2017 at 3:59 PM, Sp Pho wrote: > Hi, > > I'm creating a bridge to terminate call arriving inbound from a DID. > > Everything works fine, however the termination point plays a recorded > message - which as rightfully so by the switch is seen as early media (with > pre-answer). > > However, when the originator hangs up while the early media is playing, > the CDRs indicate NORMAL_CLEARING. If ignore early media is enabled, the > media does not play and ORIGINATOR_CANCELis the hang-up cause. > > Is there any way to receive the early media and from the CDR know whether > or not the call was subsequently answered or not by the agent (i.e. if the > originator canceled the call during the early media or while on hold)? > > Thanks in advance, > SP > -------------- next part -------------- An HTML attachment was scrubbed... URL: From khamlichi.khalil at gmail.com Wed Jun 14 12:09:34 2017 From: khamlichi.khalil at gmail.com (Khalil Khamlichi) Date: Wed, 14 Jun 2017 12:09:34 +0000 Subject: [Freeswitch-users] Callcenter module, can I originate call for an agent in uuid-standby mode ? Message-ID: Hi, I need to give my agents ability to make manual calls, hopefully without leaving their actually established call (they are in uuid-standby mode and in Idle state so there is no live member on the line ). my questions: Is it possible to originate a new call and bridge with agent uuid-standby session ? would it not break the callcenter establised uuid-standby session ? would the agent return to its uuid-standby session after the originated call is hangup ? ofcourse if this is too complicated, I would just connect the agent thru a second line, while leaving his uuid-standby call on the first line, though it would be so cool to somehow stay on that same uuid-standby session and enjoy both calllcenter module and manual dialing. Thanks in advance, and I appreciate your help. Khalil -------------- next part -------------- An HTML attachment was scrubbed... URL: From khorsmann at gmail.com Wed Jun 14 14:02:50 2017 From: khorsmann at gmail.com (Karsten Horsmann) Date: Wed, 14 Jun 2017 16:02:50 +0200 Subject: [Freeswitch-users] [SR-Users] Multiple FreeSWITCH servers behind kamailio-websocket In-Reply-To: <407012cb-dc25-b237-17d9-69462307e05b@gmail.com> References: <407012cb-dc25-b237-17d9-69462307e05b@gmail.com> Message-ID: Hello Daniel, i will try that and hopfully get an working webrtc loadbalancer in the near future with Kamailio and FreeSWITCH :). 2017-06-14 10:57 GMT+02:00 Daniel-Constantin Mierla : > Hello, > > that combination of dispatcher+websocket+xhttp modules works just fine... > > So to load balance the SIP signaling with Kamailio towards FreeSwitch, > just use the dispatcher module as usual. A sample config is available at: > > - https://www.kamailio.org/docs/modules/stable/modules/ > dispatcher.html#dispatcher.ex.config > > You need to add the support for websocket traffic via websocket module: > > - https://www.kamailio.org/docs/modules/stable/modules/ > websocket.html#idp42826164 > > or extract from the tutorial linked in a previous email on this thread. > > Cheers, > Daniel > > > On 13.06.17 16:38, Joel Serrano wrote: > > Hi Karsten, > > Have you tried with regular Kamailio (w/ dispatcher+websocket+xhttp > modules)? I don't see why it wouldn't work... > > Joel. > > > On Wed, Jun 7, 2017 at 4:47 AM, Karsten Horsmann > wrote: > >> Hello List, >> >> >> is there any howto about webrtc loadbalance in combination with kamailio >> and FreeSWITCH? >> >> I want to share one WSS address/endpoint to multiple FreeSWITCH backends. >> Or is there any other best practice? >> >> My callflow is mostly that my internal SIP Servers called my registered >> webrtc clients. >> >> Would be nice to get some input. >> >> -- >> Kind Regards >> *Karsten Horsmann* >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > _______________________________________________ > Kamailio (SER) - Users Mailing Listsr-users at lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users > > > > -- > Daniel-Constantin Mierlawww.twitter.com/miconda -- www.linkedin.com/in/miconda > Kamailio Advanced Training - www.asipto.com > Kamailio World Conference - www.kamailioworld.com > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Mit freundlichen Grüßen *Karsten Horsmann* -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian at freeswitch.org Wed Jun 14 16:26:17 2017 From: brian at freeswitch.org (Brian West) Date: Wed, 14 Jun 2017 11:26:17 -0500 Subject: [Freeswitch-users] B-Leg Early Media - Normal Clearing vs Originator Cancelled In-Reply-To: References: Message-ID: You're receiving a 487 (timeout) response. /b On Wed, Jun 14, 2017 at 5:30 AM, Sp Pho wrote: > I've pulled a log of the bad case vs good one. A diff of both can be found > here: > https://www.diffchecker.com/zWNNGibE. > > This is from the same originating number - any ideas why on line 96, > early media is detected for the good case and not the bad, would be greatly > appreciated. > > thanks in advance > > > On Tue, Jun 13, 2017 at 3:59 PM, Sp Pho wrote: > >> Hi, >> >> I'm creating a bridge to terminate call arriving inbound from a DID. >> >> Everything works fine, however the termination point plays a recorded >> message - which as rightfully so by the switch is seen as early media (with >> pre-answer). >> >> However, when the originator hangs up while the early media is playing, >> the CDRs indicate NORMAL_CLEARING. If ignore early media is enabled, the >> media does not play and ORIGINATOR_CANCELis the hang-up cause. >> >> Is there any way to receive the early media and from the CDR know whether >> or not the call was subsequently answered or not by the agent (i.e. if the >> originator canceled the call during the early media or while on hold)? >> >> Thanks in advance, >> SP >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Book a phone call (CST) Allison prompts for FreeSWITCH: *https://www.gofundme.com/allison-prompts-for-freeswitch* Got Bugs? Report them here ! | Reddit: /r/freeswitch *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian at freeswitch.org Wed Jun 14 16:27:35 2017 From: brian at freeswitch.org (Brian West) Date: Wed, 14 Jun 2017 11:27:35 -0500 Subject: [Freeswitch-users] Paging In-Reply-To: <5D4CCF18-DAA0-4F84-8203-4F69D66B251A@magicmail.mooo.com> References: <8AE4D7F5-1AE6-44F8-B85D-13C4FADC22CA@magicmail.mooo.com> <5D4CCF18-DAA0-4F84-8203-4F69D66B251A@magicmail.mooo.com> Message-ID: Without multicast you'll have to review the mad boss example in the vanilla config. Doing so in this matter doesn't scale well. ./b On Wed, Jun 14, 2017 at 6:03 AM, Rick Jarvis wrote: > Thanks Nandy. We’ve used PA for stuff, but what we’re trying to accomplish > here is paging through handsets. It’s kind of strange that there isn’t more > support for this as it’s quite a standard feature even on the old analog > PBX systems, Panasonic and so on. > > > On 14 Jun 2017, at 10:43, Nandy Dagondon wrote: > > I've done this using mod_portaudio with auto-answer from this example: > https://wiki.freeswitch.org/wiki/Mod_portaudio#PA_System_w.2F_Chime > > /nandy > > On Fri, May 19, 2017 at 12:04 PM, Lesley Pervis > wrote: > >> I've done this in Lua with outbound conference calls, but it was pretty >> complicated and you have to have endpoints that will auto-answer. >> https://freeswitch.org/confluence/display/FREESWITCH/Outboun >> d+Conference+Calls >> >> On Wed, May 17, 2017 at 10:02 AM, Rick Jarvis >> wrote: >> >>> Looking to set up paging (not multicast). What’s the best way of >>> achieving this? Specifically, I want to have the receiving handset(s) >>> answer muted for privacy reasons, so it’s literally like a PA system rather >>> than just auto answer…? >>> >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at 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 at 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 at 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 at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Book a phone call (CST) Allison prompts for FreeSWITCH: *https://www.gofundme.com/allison-prompts-for-freeswitch* Got Bugs? Report them here ! | Reddit: /r/freeswitch *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: From admin at tedssupply.com Wed Jun 14 16:36:04 2017 From: admin at tedssupply.com (admin) Date: Wed, 14 Jun 2017 12:36:04 -0400 Subject: [Freeswitch-users] When calling out , freeswitch doesn't get DTMF ? In-Reply-To: References: <15c898b5d58.2779.b07ebdf329620b8089087c7205b03f01@xbipin.com> <310cf71c.8fc3.15c8ba9c4f6.Coremail.eastour@163.com> <2a18422f.ef8.15c99b7b9bb.Coremail.eastour@163.com> <6c11dd0f.c2c1.15c9c727dc7.Coremail.eastour@163.com> <2f48505d.d9c5.15ca1d28424.Coremail.eastour@163.com> <59401B92020000310000A34D@mail.tedssupply.com> Message-ID: <59412DB4020000310000A381@mail.tedssupply.com> I don't mean to hijack the OP concern with my problem, I just wanted to reinforce I have seen this problem. And the problem is that the call originated through FS appears not to send user entered DTMF to the receiving phone. Let's help chenyzhi and then I'll take a turn. - James >>> Giovanni Maruzzelli 6/14/2017 03:53 AM >>> Never heard such problems Please pastebin your dialplan, your SIP profile, and the complete, since beginning to end, unedited, debug output of console when receiving a call which does not get DTMFs sent from mobile cell: +39 347 266 56 18 Giovanni Maruzzelli OpenTelecom.IT On Jun 13, 2017 23:07, "admin" wrote: I am encountering the same issue. I am using an ESL call to api originate, with little else except getting in and out of the port, and the caller DTMF is failing to reach the called number. The called number auto answer attendant does not respond to DTMF, and a call to a test phone confirms no DTMF. This is a new upgrade from 1.2 to 1.6 and I don't know if this was an issue in 1.2, but my users never complained before 1.6 upgrade. Ideas?... >>> chenyzhi 06/13/17 3:32 PM >>> It's a one leg call .there is no b-leg. please make a test on your freeswitch box . just type the command "originate user/1001 5000" on the freeswitch console to see if your freeswitch instance can detect dtmf input. At 2017-06-12 22:57:17, "Luis Daniel Lucio Quiroz" wrote: check if you have transcoding, and if you do, check that dftm type-codec on leg b are compatible. -- Luis Daniel Lucio Quiroz CISSP, CISM, CISA Linux, VoIP and much more fun www.okay.com.mx Need LCR? Check out LCR for FusionPBX with FreeSWITCH Need Billing? Check out Billing for FusionPBX with FreeSWITCH On Mon, Jun 12, 2017 at 9:16 AM, chenyzhi wrote: I don't think It's the DTMF type , because when I dial 5000 from x-lite (which has registered to freeswitch as 1001) ,I can hear the voice and if I press any dtmf on x-lite, freeswitch can recieve the dtmfs. This means that the DTMF type is correct ,otherwise freeswitch coudn't have received the dtmfs; when I enter the command "originate user/1001 5000" at the freeswitch console , my xlite will ring ,and I answered ,I can hear the voice ,I press some dtmf,but freeswitch can NOT receive any dtmf. really weird. At 2017-06-12 17:37:49, "Giovanni Maruzzelli" wrote: you need to check the DTMF type, you probably are using the wrong one (info-inband-rfc2833), and for some reason they are not negotiated On 12 June 2017 at 02:32, chenyzhi wrote: I have read the logs ,but I didn't find any difference. Please make a test to see if this happens in your box. At 2017-06-10 06:49:34, "David Villasmil" wrote: Have you looked at the log? Bump the logging up and see what shows up... what you're seeing is very weird David On Fri, Jun 9, 2017 at 11:18 AM chenyzhi wrote: Hi call 5000 from x-lite , you can hear the IVR voice and if you press dtmf keys , freeswitch can receive the dtmf keys . but , if you enter the command "originate user/1001 5000" , x-lite will ring ,answer it ,you can hear the IVR voice , press some keys ,the freeswitch can NOT receive any dtmf , why? _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at 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 at 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 at 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 at 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 OpenTelecom.IT cell: +39 347 266 56 18 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at 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 at 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 at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Wed Jun 14 16:41:30 2017 From: david.villasmil.work at gmail.com (David Villasmil) Date: Wed, 14 Jun 2017 16:41:30 +0000 Subject: [Freeswitch-users] When calling out , freeswitch doesn't get DTMF ? In-Reply-To: <59412DB4020000310000A381@mail.tedssupply.com> References: <15c898b5d58.2779.b07ebdf329620b8089087c7205b03f01@xbipin.com> <310cf71c.8fc3.15c8ba9c4f6.Coremail.eastour@163.com> <2a18422f.ef8.15c99b7b9bb.Coremail.eastour@163.com> <6c11dd0f.c2c1.15c9c727dc7.Coremail.eastour@163.com> <2f48505d.d9c5.15ca1d28424.Coremail.eastour@163.com> <59401B92020000310000A34D@mail.tedssupply.com> <59412DB4020000310000A381@mail.tedssupply.com> Message-ID: I do this on a daily basis, originate and send to an ivr to get dtmf digits, no issues ever found... On Wed, Jun 14, 2017 at 6:36 PM admin wrote: > I don't mean to hijack the OP concern with my problem, I just wanted to > reinforce I have seen this problem. And the problem is that the call > originated through FS appears not to send user entered DTMF to the > receiving phone. Let's help chenyzhi and then I'll take a turn. > > - James > > >>> Giovanni Maruzzelli 6/14/2017 03:53 AM >>> > > Never heard such problems > > > Please pastebin your dialplan, your SIP profile, and the complete, since > beginning to end, unedited, debug output of console when receiving a call > which does not get DTMFs > > > > > sent from mobile > cell: +39 347 266 56 18 > Giovanni Maruzzelli > OpenTelecom.IT > > > On Jun 13, 2017 23:07, "admin" wrote: > >> I am encountering the same issue. I am using an ESL call to api >> originate, with little else except getting in and out of the port, and the >> caller DTMF is failing to reach the called number. The called number auto >> answer attendant does not respond to DTMF, and a call to a test phone >> confirms no DTMF. This is a new upgrade from 1.2 to 1.6 and I don't know if >> this was an issue in 1.2, but my users never complained before 1.6 upgrade. >> Ideas?... >> >> >> >> >>> chenyzhi 06/13/17 3:32 PM >>> >> >> It's a one leg call .there is no b-leg. >> >> >> please make a test on your freeswitch box . >> >> >> just type the command "originate user/1001 5000" on the freeswitch >> console to see if your freeswitch instance can detect dtmf input. >> >> >> >> >> >> >> >> >> At 2017-06-12 22:57:17, "Luis Daniel Lucio Quiroz" < >> luis.daniel.lucio at gmail.com> wrote: >> >> check if you have transcoding, and if you do, check that dftm type-codec >> on leg b are compatible. >> >> >> -- >> >> Luis Daniel Lucio Quiroz >> CISSP, CISM, CISA >> Linux, VoIP and much more fun >> www.okay.com.mx >> >> Need LCR? Check out LCR for FusionPBX with FreeSWITCH >> Need Billing? Check out Billing for FusionPBX with FreeSWITCH >> >> >> On Mon, Jun 12, 2017 at 9:16 AM, chenyzhi wrote: >> >>> I don't think It's the DTMF type , because when I dial 5000 from x-lite >>> (which has registered to freeswitch as 1001) ,I can hear the voice and if I >>> press any dtmf on x-lite, freeswitch can recieve the dtmfs. This means that >>> the DTMF type is correct ,otherwise freeswitch coudn't have received the >>> dtmfs; >>> >>> >>> when I enter the command "originate user/1001 5000" at the freeswitch >>> console , my xlite will ring ,and I answered ,I can hear the voice ,I press >>> some dtmf,but freeswitch can NOT receive any dtmf. really weird. >>> >>> >>> >>> >>> >>> >>> >>> >>> At 2017-06-12 17:37:49, "Giovanni Maruzzelli" wrote: >>> >>> you need to check the DTMF type, you probably are using the wrong one >>> (info-inband-rfc2833), and for some reason they are not negotiated >>> >>> >>> On 12 June 2017 at 02:32, chenyzhi wrote: >>> >>>> I have read the logs ,but I didn't find any difference. >>>> >>>> >>>> Please make a test to see if this happens in your box. >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> At 2017-06-10 06:49:34, "David Villasmil" < >>>> david.villasmil.work at gmail.com> wrote: >>>> >>>> Have you looked at the log? Bump the logging up and see what shows >>>> up... what you're seeing is very weird >>>> >>>> David >>>> >>>> On Fri, Jun 9, 2017 at 11:18 AM chenyzhi wrote: >>>> >>>>> Hi >>>>> >>>>> >>>>> call 5000 from x-lite , you can hear the IVR voice and if you press >>>>> dtmf keys , freeswitch can receive the dtmf keys . >>>>> >>>>> >>>>> but , if you enter the command "originate user/1001 5000" , x-lite >>>>> will ring ,answer it ,you can hear the IVR voice , press some keys ,the >>>>> freeswitch can NOT receive any dtmf , >>>>> >>>>> >>>>> why? >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at 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 at 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 at 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 at 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 >>> OpenTelecom.IT >>> cell: +39 347 266 56 18 >>> >>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at 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 at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From gmaruzz at gmail.com Wed Jun 14 16:42:11 2017 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Wed, 14 Jun 2017 18:42:11 +0200 Subject: [Freeswitch-users] When calling out , freeswitch doesn't get DTMF ? In-Reply-To: <59412DB4020000310000A381@mail.tedssupply.com> References: <15c898b5d58.2779.b07ebdf329620b8089087c7205b03f01@xbipin.com> <310cf71c.8fc3.15c8ba9c4f6.Coremail.eastour@163.com> <2a18422f.ef8.15c99b7b9bb.Coremail.eastour@163.com> <6c11dd0f.c2c1.15c9c727dc7.Coremail.eastour@163.com> <2f48505d.d9c5.15ca1d28424.Coremail.eastour@163.com> <59401B92020000310000A34D@mail.tedssupply.com> <59412DB4020000310000A381@mail.tedssupply.com> Message-ID: On 14 June 2017 at 18:36, admin wrote: > I don't mean to hijack the OP concern with my problem, I just wanted to > reinforce I have seen this problem. And the problem is that the call > originated through FS appears not to send user entered DTMF to the > receiving phone. Let's help chenyzhi and then I'll take a turn. > I just tested right now from console with: bgapi originate user/1011 5000 and originate user/1011 5000 and it works Please pastebin your FreeSWITCH version (eg, type "version" in console), your dialplan, your SIP profile, and the complete, since beginning to end, unedited, debug output of console when receiving a call which does not get DTMFs Maybe helping you we'll help him too > > - James > > >>> Giovanni Maruzzelli 6/14/2017 03:53 AM >>> > > Never heard such problems > > > Please pastebin your dialplan, your SIP profile, and the complete, since > beginning to end, unedited, debug output of console when receiving a call > which does not get DTMFs > > > > > sent from mobile > cell: +39 347 266 56 18 > Giovanni Maruzzelli > OpenTelecom.IT > > > On Jun 13, 2017 23:07, "admin" wrote: > >> I am encountering the same issue. I am using an ESL call to api >> originate, with little else except getting in and out of the port, and the >> caller DTMF is failing to reach the called number. The called number auto >> answer attendant does not respond to DTMF, and a call to a test phone >> confirms no DTMF. This is a new upgrade from 1.2 to 1.6 and I don't know if >> this was an issue in 1.2, but my users never complained before 1.6 upgrade. >> Ideas?... >> >> >> >> >>> chenyzhi 06/13/17 3:32 PM >>> >> >> It's a one leg call .there is no b-leg. >> >> >> please make a test on your freeswitch box . >> >> >> just type the command "originate user/1001 5000" on the freeswitch >> console to see if your freeswitch instance can detect dtmf input. >> >> >> >> >> >> >> >> >> At 2017-06-12 22:57:17, "Luis Daniel Lucio Quiroz" < >> luis.daniel.lucio at gmail.com> wrote: >> >> check if you have transcoding, and if you do, check that dftm type-codec >> on leg b are compatible. >> >> >> -- >> >> Luis Daniel Lucio Quiroz >> CISSP, CISM, CISA >> Linux, VoIP and much more fun >> www.okay.com.mx >> >> Need LCR? Check out LCR for FusionPBX with FreeSWITCH >> Need Billing? Check out Billing for FusionPBX with FreeSWITCH >> >> >> On Mon, Jun 12, 2017 at 9:16 AM, chenyzhi wrote: >> >>> I don't think It's the DTMF type , because when I dial 5000 from x-lite >>> (which has registered to freeswitch as 1001) ,I can hear the voice and if I >>> press any dtmf on x-lite, freeswitch can recieve the dtmfs. This means that >>> the DTMF type is correct ,otherwise freeswitch coudn't have received the >>> dtmfs; >>> >>> >>> when I enter the command "originate user/1001 5000" at the freeswitch >>> console , my xlite will ring ,and I answered ,I can hear the voice ,I press >>> some dtmf,but freeswitch can NOT receive any dtmf. really weird. >>> >>> >>> >>> >>> >>> >>> >>> >>> At 2017-06-12 17:37:49, "Giovanni Maruzzelli" wrote: >>> >>> you need to check the DTMF type, you probably are using the wrong one >>> (info-inband-rfc2833), and for some reason they are not negotiated >>> >>> >>> On 12 June 2017 at 02:32, chenyzhi wrote: >>> >>>> I have read the logs ,but I didn't find any difference. >>>> >>>> >>>> Please make a test to see if this happens in your box. >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> At 2017-06-10 06:49:34, "David Villasmil" < >>>> david.villasmil.work at gmail.com> wrote: >>>> >>>> Have you looked at the log? Bump the logging up and see what shows >>>> up... what you're seeing is very weird >>>> >>>> David >>>> >>>> On Fri, Jun 9, 2017 at 11:18 AM chenyzhi wrote: >>>> >>>>> Hi >>>>> >>>>> >>>>> call 5000 from x-lite , you can hear the IVR voice and if you press >>>>> dtmf keys , freeswitch can receive the dtmf keys . >>>>> >>>>> >>>>> but , if you enter the command "originate user/1001 5000" , x-lite >>>>> will ring ,answer it ,you can hear the IVR voice , press some keys ,the >>>>> freeswitch can NOT receive any dtmf , >>>>> >>>>> >>>>> why? >>>>> >>>>> >>>>> >>>>> >>>>> ____________________________________________________________ >>>>> _____________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at 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 at 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 at 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 at 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 >>> OpenTelecom.IT >>> cell: +39 347 266 56 18 >>> >>> >>> >>> >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at 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 at 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 at 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 at 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 at 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 at 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 OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: From rick at magicmail.mooo.com Wed Jun 14 16:53:08 2017 From: rick at magicmail.mooo.com (Rick Jarvis) Date: Wed, 14 Jun 2017 17:53:08 +0100 Subject: [Freeswitch-users] Paging In-Reply-To: References: <8AE4D7F5-1AE6-44F8-B85D-13C4FADC22CA@magicmail.mooo.com> <5D4CCF18-DAA0-4F84-8203-4F69D66B251A@magicmail.mooo.com> Message-ID: I’m correct in thinking that for multicast to work, the endpoints have to be on the same LAN, am I? Haven’t ever played with multicast. I don’t know why I hadn’t realised that mad boss looks as if it will do exactly what I’m after, hopefully good for a dozen extensions or so… and it mutes all the handsets in the group it calls, right? > On 14 Jun 2017, at 17:27, Brian West wrote: > > Without multicast you'll have to review the mad boss example in the vanilla config. Doing so in this matter doesn't scale well. > > ./b > > On Wed, Jun 14, 2017 at 6:03 AM, Rick Jarvis > wrote: > Thanks Nandy. We’ve used PA for stuff, but what we’re trying to accomplish here is paging through handsets. It’s kind of strange that there isn’t more support for this as it’s quite a standard feature even on the old analog PBX systems, Panasonic and so on. > > >> On 14 Jun 2017, at 10:43, Nandy Dagondon > wrote: >> >> I've done this using mod_portaudio with auto-answer from this example: >> https://wiki.freeswitch.org/wiki/Mod_portaudio#PA_System_w.2F_Chime >> >> /nandy >> >> On Fri, May 19, 2017 at 12:04 PM, Lesley Pervis > wrote: >> I've done this in Lua with outbound conference calls, but it was pretty complicated and you have to have endpoints that will auto-answer. https://freeswitch.org/confluence/display/FREESWITCH/Outbound+Conference+Calls >> >> On Wed, May 17, 2017 at 10:02 AM, Rick Jarvis > wrote: >> Looking to set up paging (not multicast). What’s the best way of achieving this? Specifically, I want to have the receiving handset(s) answer muted for privacy reasons, so it’s literally like a PA system rather than just auto answer…? >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at 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 at 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 at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -- > Brian West > brian at freeswitch.org > Twitter: @FreeSWITCH , @briankwest > > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > Book a phone call (CST) > > Allison prompts for FreeSWITCH: > > https://www.gofundme.com/allison-prompts-for-freeswitch > Got Bugs? Report them here ! | Reddit: /r/freeswitch > T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378) > Skype:briankwest > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From jnankin at gmail.com Wed Jun 14 17:34:50 2017 From: jnankin at gmail.com (Josh Nankin) Date: Wed, 14 Jun 2017 12:34:50 -0500 Subject: [Freeswitch-users] hello Message-ID: -------------- next part -------------- An HTML attachment was scrubbed... URL: From jnankin at gmail.com Wed Jun 14 19:22:15 2017 From: jnankin at gmail.com (Josh Nankin) Date: Wed, 14 Jun 2017 14:22:15 -0500 Subject: [Freeswitch-users] RPORT still being sent in TCP calls In-Reply-To: References: <8E832DB1-BEF7-4EF1-826F-ED57C966E794@jerris.com> Message-ID: Opened https://freeswitch.org/jira/browse/FS-10392 with regards to this On Wed, Jun 14, 2017 at 1:22 PM, Daniel Greenwald wrote: > > ---------- Forwarded message ---------- > From: Michael Jerris > Date: Mon, Jun 12, 2017 at 4:42 PM > Subject: Re: [Freeswitch-users] RPORT still being sent in TCP calls > To: FreeSWITCH Users Help > > > Change is still in there, confirmed. The packages you are using are very > old (but should also have that patch in it)… I see that you are supplying > modified sip traces. If you can reproduce this on master code, please > create a Jira with configuration and full debug logs with sip trace > (unmodified) attached. > > Thanks > Mike > > On Jun 12, 2017, at 4:32 PM, Daniel Greenwald wrote: > > We have upgraded to latest STABLE from Ubuntu packages but we are still > seeing rport in TCP calls: > > > INVITE sip:xxxxxxx at sip.freeswitch.com;transport=tcp SIP/2.0 > Via: SIP/2.0/TCP x.x.x.x;rport;branch=z9hG4bKZDKF5N8QcFZcH > Max-Forwards: 70 > From: "user" ;tag=Z1pQm54mB22De > To: > Call-ID: c79656fc-d2d0-4446-90c3-060dabf82fd6 > CSeq: 108312748 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.6.10-4-726448d~64bit > > On Fri, Jun 9, 2017 at 2:56 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> Odd number releases are not stable releases they are dev releases. >> The version you quoted is from may 2015 2 years ago. >> >> The latest stable release is 1.6.17 and latest build release changes >> every day. >> >> >> >> On Fri, Jun 9, 2017 at 10:50 AM, Daniel Greenwald >> wrote: >> >>> We have noticed that FS is sending RPORT in TCP calls to a gateway. It >>> was reported as fixed in this bug: >>> https://freeswitch.org/jira/browse/FS-6612 >>> >>> We are running: >>> 1.5.15b+git~20150512T053645Z~9eb887af47~64bit >>> >>> I am not sure why RPORT is still being sent. Is this there a config >>> parameter which needs to be set to suppress the RPORT? Or was this change >>> reverted in later versions for some reason. Provider is telling us we >>> should not be sending RPORT in TCP... >>> >>> Any info would be greatly appreciated. >>> >>> Thanks! >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at 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 at 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 ♬ @anthmfs ♬ @FreeSWITCH ♬ >> >> ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ >> http://twitter.com/FreeSWITCH >> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ >> * >> >> ClueCon Weekly Development Call >> ☎ sip:888 at conference.freeswitch.org ☎ +19193869900 <(919)%20386-9900> >> >> https://www.youtube.com/watch?v=9XXgW34t40s >> https://www.youtube.com/watch?v=NLaDpGQuZDA >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From prex5609 at gmail.com Wed Jun 14 23:07:59 2017 From: prex5609 at gmail.com (Peter Rex) Date: Wed, 14 Jun 2017 17:07:59 -0600 Subject: [Freeswitch-users] Paging In-Reply-To: References: <8AE4D7F5-1AE6-44F8-B85D-13C4FADC22CA@magicmail.mooo.com> <5D4CCF18-DAA0-4F84-8203-4F69D66B251A@magicmail.mooo.com> Message-ID: I thin I added "flags{mute} inline" to each leg dialstring. On Wed, Jun 14, 2017 at 10:53 AM, Rick Jarvis wrote: > I’m correct in thinking that for multicast to work, the endpoints have to > be on the same LAN, am I? Haven’t ever played with multicast. > > I don’t know why I hadn’t realised that mad boss looks as if it will do > exactly what I’m after, hopefully good for a dozen extensions or so… and it > mutes all the handsets in the group it calls, right? > > > On 14 Jun 2017, at 17:27, Brian West wrote: > > Without multicast you'll have to review the mad boss example in the > vanilla config. Doing so in this matter doesn't scale well. > > ./b > > On Wed, Jun 14, 2017 at 6:03 AM, Rick Jarvis > wrote: > >> Thanks Nandy. We’ve used PA for stuff, but what we’re trying to >> accomplish here is paging through handsets. It’s kind of strange that there >> isn’t more support for this as it’s quite a standard feature even on the >> old analog PBX systems, Panasonic and so on. >> >> >> On 14 Jun 2017, at 10:43, Nandy Dagondon wrote: >> >> I've done this using mod_portaudio with auto-answer from this example: >> https://wiki.freeswitch.org/wiki/Mod_portaudio#PA_System_w.2F_Chime >> >> /nandy >> >> On Fri, May 19, 2017 at 12:04 PM, Lesley Pervis >> wrote: >> >>> I've done this in Lua with outbound conference calls, but it was pretty >>> complicated and you have to have endpoints that will auto-answer. >>> https://freeswitch.org/confluence/display/FREESWITCH/Outboun >>> d+Conference+Calls >>> >>> On Wed, May 17, 2017 at 10:02 AM, Rick Jarvis >>> wrote: >>> >>>> Looking to set up paging (not multicast). What’s the best way of >>>> achieving this? Specifically, I want to have the receiving handset(s) >>>> answer muted for privacy reasons, so it’s literally like a PA system rather >>>> than just auto answer…? >>>> >>>> >>>> ____________________________________________________________ >>>> _____________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at 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 at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>> switch-users >>>> http://www.freeswitch.org >>> >>> >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at 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 at 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 at 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 at 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 at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > > *Brian West* > brian at freeswitch.org > > *Twitter: @FreeSWITCH , @briankwest* > > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > Book a phone call (CST) > > Allison prompts for FreeSWITCH: > > *https://www.gofundme.com/allison-prompts-for-freeswitch* > > > Got Bugs? Report them here ! | Reddit: > /r/freeswitch > > *T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002 <(918)%20420-9002> > | *M:*+1918424WEST (9378) > *Skype:*briankwest > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From agubbe at gmail.com Thu Jun 15 08:07:01 2017 From: agubbe at gmail.com (=?UTF-8?Q?Agust=C3=AD_Ubalde_Bellot?=) Date: Thu, 15 Jun 2017 10:07:01 +0200 Subject: [Freeswitch-users] Freeswitch sslv3 support In-Reply-To: References: Message-ID: Hi Brian, Is possible to disable for web socket secure connections too? Thanks, Agustí 2017-06-13 13:24 GMT+02:00 Agustí Ubalde Bellot : > Hi all, > > Is there a FreeSWITCH update where sslv3 support is disabled? > > > Thanks, > Agustí > -------------- next part -------------- An HTML attachment was scrubbed... URL: From eastour at 163.com Thu Jun 15 02:28:43 2017 From: eastour at 163.com (chenyzhi) Date: Thu, 15 Jun 2017 10:28:43 +0800 (CST) Subject: [Freeswitch-users] When calling out , freeswitch doesn't get DTMF ? In-Reply-To: References: <15c898b5d58.2779.b07ebdf329620b8089087c7205b03f01@xbipin.com> <310cf71c.8fc3.15c8ba9c4f6.Coremail.eastour@163.com> <2a18422f.ef8.15c99b7b9bb.Coremail.eastour@163.com> <6c11dd0f.c2c1.15c9c727dc7.Coremail.eastour@163.com> <2f48505d.d9c5.15ca1d28424.Coremail.eastour@163.com> <59401B92020000310000A34D@mail.tedssupply.com> <59412DB4020000310000A381@mail.tedssupply.com> Message-ID: <7d1e3f12.4750.15ca994d89d.Coremail.eastour@163.com> the version is: FreeSWITCH Version 1.9.0+git~20170518T231917Z~a1fc18aee5~32bit (git a1fc18a 2017-05-18 23:19:17Z 32bit) the complete, since beginning to end, unedited, debug output of console when making a outgoing call which does not get DTMFs and the whole conf folder is in the attatchment. thx! 在 2017-06-15 00:42:11，"Giovanni Maruzzelli" 写道： On 14 June 2017 at 18:36, admin wrote: I don't mean to hijack the OP concern with my problem, I just wanted to reinforce I have seen this problem. And the problem is that the call originated through FS appears not to send user entered DTMF to the receiving phone. Let's help chenyzhi and then I'll take a turn. I just tested right now from console with: bgapi originate user/1011 5000 and originate user/1011 5000 and it works Please pastebin your FreeSWITCH version (eg, type "version" in console), your dialplan, your SIP profile, and the complete, since beginning to end, unedited, debug output of console when receiving a call which does not get DTMFs Maybe helping you we'll help him too - James >>> Giovanni Maruzzelli 6/14/2017 03:53 AM >>> Never heard such problems Please pastebin your dialplan, your SIP profile, and the complete, since beginning to end, unedited, debug output of console when receiving a call which does not get DTMFs sent from mobile cell: +39 347 266 56 18 Giovanni Maruzzelli OpenTelecom.IT On Jun 13, 2017 23:07, "admin" wrote: I am encountering the same issue. I am using an ESL call to api originate, with little else except getting in and out of the port, and the caller DTMF is failing to reach the called number. The called number auto answer attendant does not respond to DTMF, and a call to a test phone confirms no DTMF. This is a new upgrade from 1.2 to 1.6 and I don't know if this was an issue in 1.2, but my users never complained before 1.6 upgrade. Ideas?... >>> chenyzhi 06/13/17 3:32 PM >>> It's a one leg call .there is no b-leg. please make a test on your freeswitch box . just type the command "originate user/1001 5000" on the freeswitch console to see if your freeswitch instance can detect dtmf input. At 2017-06-12 22:57:17, "Luis Daniel Lucio Quiroz" wrote: check if you have transcoding, and if you do, check that dftm type-codec on leg b are compatible. -- Luis Daniel Lucio Quiroz CISSP, CISM, CISA Linux, VoIP and much more fun www.okay.com.mx Need LCR? Check out LCR for FusionPBX with FreeSWITCH Need Billing? Check out Billing for FusionPBX with FreeSWITCH On Mon, Jun 12, 2017 at 9:16 AM, chenyzhi wrote: I don't think It's the DTMF type , because when I dial 5000 from x-lite (which has registered to freeswitch as 1001) ,I can hear the voice and if I press any dtmf on x-lite, freeswitch can recieve the dtmfs. This means that the DTMF type is correct ,otherwise freeswitch coudn't have received the dtmfs; when I enter the command "originate user/1001 5000" at the freeswitch console , my xlite will ring ,and I answered ,I can hear the voice ,I press some dtmf,but freeswitch can NOT receive any dtmf. really weird. At 2017-06-12 17:37:49, "Giovanni Maruzzelli" wrote: you need to check the DTMF type, you probably are using the wrong one (info-inband-rfc2833), and for some reason they are not negotiated On 12 June 2017 at 02:32, chenyzhi wrote: I have read the logs ,but I didn't find any difference. Please make a test to see if this happens in your box. At 2017-06-10 06:49:34, "David Villasmil" wrote: Have you looked at the log? Bump the logging up and see what shows up... what you're seeing is very weird David On Fri, Jun 9, 2017 at 11:18 AM chenyzhi wrote: Hi call 5000 from x-lite , you can hear the IVR voice and if you press dtmf keys , freeswitch can receive the dtmf keys . but , if you enter the command "originate user/1001 5000" , x-lite will ring ,answer it ,you can hear the IVR voice , press some keys ,the freeswitch can NOT receive any dtmf , why? _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at 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 at 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 at 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 at 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 OpenTelecom.IT cell: +39 347 266 56 18 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at 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 at 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 at 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 at 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 at 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 at 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 OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: freeswitch-log-4callingoutdoesnotgetdtmf.7z Type: application/x-7z-compressed Size: 99772 bytes Desc: not available URL: From ron.menez at entropysolution.com Thu Jun 15 07:49:04 2017 From: ron.menez at entropysolution.com (Ron) Date: Thu, 15 Jun 2017 07:49:04 +0000 Subject: [Freeswitch-users] AMR Codec for Audio File Playback Message-ID: <34D864D4-4CD0-4F6D-8B3C-83F359411180@entropysolution.com> Hi All, Is it possible to execute a command playback for an audio file using AMR Codec? We tried the following configuration in dialplan and gave different errors: Configuration with “pre-answer”: Error Log: EXECUTE sofia/internal/09570000001 at 192.168.1.129:5062 pre_answer() 2017-06-15 15:39:45.285763 [INFO] mod_dptools.c:1355 Sending early media 2017-06-15 15:39:45.285763 [DEBUG] switch_core_media.c:3056 Set Codec sofia/internal/09570000001 at 192.168.1.129:5062 AMR/0 0 ms 160 samples 12200 bits 1 channels 2017-06-15 15:39:45.285763 [DEBUG] switch_core_codec.c:111 sofia/internal/09570000001 at 192.168.1.129:5062 Original read codec set to AMR:96 2017-06-15 15:39:45.285763 [DEBUG] switch_core_media.c:6927 PROXY AUDIO RTP [sofia/internal/09570000001 at 192.168.1.129:5062] 192.168.1.129:62020->192.168.1.129:62020 codec: 98 ms: 20 2017-06-15 15:39:45.285763 [ERR] switch_core_media.c:7549 AUDIO RTP REPORTS ERROR: [Missing local host] 2017-06-15 15:39:45.285763 [NOTICE] switch_core_media.c:7550 Hangup sofia/internal/09570000001 at 192.168.1.129:5062 [CS_EXECUTE] [DESTINATION_OUT_OF_ORDER] Configuration with “answer”: Error Log: 2017-06-15 15:41:49.965732 [NOTICE] mod_dptools.c:1312 Channel [sofia/internal/09570000001 at 192.168.1.129:5062] has been answered 2017-06-15 15:41:49.965732 [DEBUG] switch_channel.c:3772 (sofia/internal/09570000001 at 192.168.1.129:5062) Callstate Change EARLY -> ACTIVE 2017-06-15 15:41:49.965732 [DEBUG] sofia.c:7048 Channel sofia/internal/09570000001 at 192.168.1.129:5062 entering state [completed][200] 2017-06-15 15:41:49.965732 [DEBUG] sofia.c:7048 Channel sofia/internal/09570000001 at 192.168.1.129:5062 entering state [ready][200] EXECUTE sofia/internal/09570000001 at 192.168.1.129:5062 sleep(2000) 2017-06-15 15:41:49.985728 [DEBUG] switch_rtp.c:7247 Correct audio ip/port confirmed. EXECUTE sofia/internal/09570000001 at 192.168.1.129:5062 playback(/usr/local/freeswitch/sounds/aaaaa.wav) 2017-06-15 15:41:51.985732 [DEBUG] switch_core_file.c:342 File /usr/local/freeswitch/sounds/aaaaa.wav sample rate 44100 doesn't match requested rate 8000 2017-06-15 15:41:51.985732 [WARNING] switch_core_file.c:360 File has 2 channels, muxing to 1 channel will occur. 2017-06-15 15:41:51.985732 [DEBUG] switch_ivr_play_say.c:1498 Codec Activated L16 at 8000hz 1 channels 20ms 2017-06-15 15:41:52.005728 [ERR] mod_amr.c:338 This codec is only usable in passthrough mode! 2017-06-15 15:41:52.005728 [ERR] switch_core_io.c:1434 Codec AMR encoder error! 2017-06-15 15:41:52.005728 [DEBUG] switch_ivr_play_say.c:1942 done playing file /usr/local/freeswitch/sounds/aaaaa.wav 2017-06-15 15:41:52.005728 [NOTICE] switch_core_state_machine.c:385 sofia/internal/09570000001 at 192.168.1.129:5062 has executed the last dialplan instruction, hanging up. 2017-06-15 15:41:52.005728 [NOTICE] switch_core_state_machine.c:387 Hangup sofia/internal/09570000001 at 192.168.1.129:5062 [CS_EXECUTE] [NORMAL_CLEARING] We also tried to use the “fs_encode" and tried AMR codec to encode the wav file but we received the error below: Opening file aaaaa.wav Opening file aaaaa.AMR 2017-05-25 16:18:32.078353 [INFO] mod_native_file.c:101 Opening File [aaaaa.AMR] 8000hz Frame size is 160 2017-05-25 16:18:32.078366 [ERR] mod_amr.c:338 This codec is only usable in passthrough mode! Codec encoder error 2017-05-25 16:18:32.078372 [WARNING] switch_core_codec.c:920 Codec is not initialized! We tried using the native file configuration and absolute codec but same errors shown for “pre_answer” and “answer” configuration. May we request your help if there is another way to run playback for audio files using AMR codec. Thank you. Best Regard, Ron Menez ron.menez at entropysolution.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From USZPELSV at comunycarse.com Thu Jun 15 07:49:24 2017 From: USZPELSV at comunycarse.com (Sven Uszpelkat) Date: Thu, 15 Jun 2017 07:49:24 +0000 Subject: [Freeswitch-users] Call recording Message-ID: <7c47071f86ce436186b4ab8bf58de46c@w12mad02.comunycarse.com> Hello, We are using FreeSWITCH as a third-party recording application, i.e. we are receiving SIP calls with the complete audio of conversations taking place on another switch and we are saving this audio to a file. To achieve this we are using a simple script similar to this: session:answer() while(session:ready() == true) do test = session:recordFile("/usr/local/freeswitch/recordings/test.wav", 18000, 0, 300) session:setAutoHangup(false) session:hangup() end This script will be invoked by the following dialplan: Basically it seems to work quite well, but sometimes there are missing audio at the end of the recorded file. Usually it's only a few seconds, but sometimes it seems to be more. (It's like the recording sometimes goes behind the real call and when the hangup event is received the remaining audio is discarded.) What could be the reason for this behavior? Is there something wrong with the script or is there a better way to achieve our goal? Many thanks in advance. Best regards, Sven Uszpelkat Departamento I+D Comunycarse Network Consultants, S.L. [Descripción: Descripción: http://www.comunycarse.com/email_images/facebook_16.jpg] [Descripción: Descripción: http://www.comunycarse.com/email_images/linkedin_16.jpg] [Descripción: Descripción: http://www.comunycarse.com/email_images/twitter_16.jpg] [Descripción: Descripción: http://www.comunycarse.com/email_images/wordpress_16.jpg] Joaquín Turina, 2 28224 Pozuelo de Alarcón MADRID Tlf. +34 917 498 700 Fax +34 917 498 720 Sabino Arana, 18 08028 BARCELONA Tlf. +34 934 098 480 Fax +34 934 098 490 http://www.comunycarse.com AVISO LEGAL La presente comunicación y sus anexos tiene como destinatario la persona a la que va dirigida, por lo que si usted lo recibe por error debe notificarlo al remitente y eliminarlo de su sistema, no pudiendo utilizarlo, total o parcialmente, para ningún fin. Su contenido puede tener información confidencial o protegida legalmente y únicamente expresa la opinión del remitente. El uso del correo electrónico vía internet no permite asegurar ni la confidencialidad de los mensajes ni su correcta recepción. En el caso de que el destinatario no consintiera la utilización del correo electrónico deberá ponerlo en nuestro conocimiento inmediatamente. DISCLAIMER This message and its attachments are intended exclusively for the named addressee. If you receive this message by mistake, please delete it immediately from your system and notify the sender. You may not use this message or any part of it for any purpose. The message may contain information that is confidential or protected by law, and any opinions expressed are those of the individual sender. Internet email guarantees neither the confidentiality nor the proper receipt of the message sent. If the addressee of this message does not consent to the use of internet e-mail, please inform us immediately. -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 1630 bytes Desc: image001.jpg URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image002.jpg Type: image/jpeg Size: 1694 bytes Desc: image002.jpg URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image003.jpg Type: image/jpeg Size: 1628 bytes Desc: image003.jpg URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image004.jpg Type: image/jpeg Size: 1402 bytes Desc: image004.jpg URL: From kkothari157 at gmail.com Thu Jun 15 06:48:20 2017 From: kkothari157 at gmail.com (Ketan Kothari) Date: Thu, 15 Jun 2017 12:18:20 +0530 Subject: [Freeswitch-users] Verto failover Message-ID: Can we setup Verto communicator *User-Interface* and *FreeSWITCH* on separate server? If yes *--->* We have 2 FreeSWITCH servers and 1 User-Interface server. So how to load balance or fail-over of Verto? -------------- next part -------------- An HTML attachment was scrubbed... URL: From gmaruzz at gmail.com Thu Jun 15 09:04:48 2017 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Thu, 15 Jun 2017 11:04:48 +0200 Subject: [Freeswitch-users] When calling out , freeswitch doesn't get DTMF ? In-Reply-To: <7d1e3f12.4750.15ca994d89d.Coremail.eastour@163.com> References: <15c898b5d58.2779.b07ebdf329620b8089087c7205b03f01@xbipin.com> <310cf71c.8fc3.15c8ba9c4f6.Coremail.eastour@163.com> <2a18422f.ef8.15c99b7b9bb.Coremail.eastour@163.com> <6c11dd0f.c2c1.15c9c727dc7.Coremail.eastour@163.com> <2f48505d.d9c5.15ca1d28424.Coremail.eastour@163.com> <59401B92020000310000A34D@mail.tedssupply.com> <59412DB4020000310000A381@mail.tedssupply.com> <7d1e3f12.4750.15ca994d89d.Coremail.eastour@163.com> Message-ID: On 15 June 2017 at 04:28, chenyzhi wrote: > the version is: FreeSWITCH Version 1.9.0+git~20170518T231917Z~a1fc18aee5~32bit > (git a1fc18a 2017-05-18 23:19:17Z 32bit) > > the complete, since beginning to end, unedited, debug output of console > when making a outgoing call which does not get DTMFs > > and the whole conf folder is in the attatchment. > > from the log, seems it does not read all the IVR messages correctly to you, it exits straight away... are you able to correctly hear all the IVR messages? also, can you take a SIP trace? (from console: "sofia global siptrace on") I suspect you have a NAT problem of some sort Also, I see you are on MASTER git, on Windows, and on 32 bit... Not sure this is supported... Have you has this problems with stable branch (1.6.x)? > thx! > > > > > > 在 2017-06-15 00:42:11，"Giovanni Maruzzelli" 写道： > > > > On 14 June 2017 at 18:36, admin wrote: > >> I don't mean to hijack the OP concern with my problem, I just wanted to >> reinforce I have seen this problem. And the problem is that the call >> originated through FS appears not to send user entered DTMF to the >> receiving phone. Let's help chenyzhi and then I'll take a turn. >> > > I just tested right now from console with: > > > bgapi originate user/1011 5000 > > and > > originate user/1011 5000 > > > and it works > > > Please pastebin your FreeSWITCH version (eg, type "version" in console), > your dialplan, your SIP profile, and the complete, since beginning to end, > unedited, debug output of console when receiving a call which does not get > DTMFs > > > Maybe helping you we'll help him too > > >> >> - James >> >> >>> Giovanni Maruzzelli 6/14/2017 03:53 AM >>> >> >> Never heard such problems >> >> >> Please pastebin your dialplan, your SIP profile, and the complete, since >> beginning to end, unedited, debug output of console when receiving a call >> which does not get DTMFs >> >> >> >> >> sent from mobile >> cell: +39 347 266 56 18 >> Giovanni Maruzzelli >> OpenTelecom.IT >> >> >> On Jun 13, 2017 23:07, "admin" wrote: >> >>> I am encountering the same issue. I am using an ESL call to api >>> originate, with little else except getting in and out of the port, and the >>> caller DTMF is failing to reach the called number. The called number auto >>> answer attendant does not respond to DTMF, and a call to a test phone >>> confirms no DTMF. This is a new upgrade from 1.2 to 1.6 and I don't know if >>> this was an issue in 1.2, but my users never complained before 1.6 upgrade. >>> Ideas?... >>> >>> >>> >>> >>> chenyzhi 06/13/17 3:32 PM >>> >>> >>> It's a one leg call .there is no b-leg. >>> >>> >>> please make a test on your freeswitch box . >>> >>> >>> just type the command "originate user/1001 5000" on the freeswitch >>> console to see if your freeswitch instance can detect dtmf input. >>> >>> >>> >>> >>> >>> >>> >>> >>> At 2017-06-12 22:57:17, "Luis Daniel Lucio Quiroz" < >>> luis.daniel.lucio at gmail.com> wrote: >>> >>> check if you have transcoding, and if you do, check that dftm type-codec >>> on leg b are compatible. >>> >>> >>> -- >>> >>> Luis Daniel Lucio Quiroz >>> CISSP, CISM, CISA >>> Linux, VoIP and much more fun >>> www.okay.com.mx >>> >>> Need LCR? Check out LCR for FusionPBX with FreeSWITCH >>> Need Billing? Check out Billing for FusionPBX with FreeSWITCH >>> >>> >>> On Mon, Jun 12, 2017 at 9:16 AM, chenyzhi wrote: >>> >>>> I don't think It's the DTMF type , because when I dial 5000 from x-lite >>>> (which has registered to freeswitch as 1001) ,I can hear the voice and if I >>>> press any dtmf on x-lite, freeswitch can recieve the dtmfs. This means that >>>> the DTMF type is correct ,otherwise freeswitch coudn't have received the >>>> dtmfs; >>>> >>>> >>>> when I enter the command "originate user/1001 5000" at the freeswitch >>>> console , my xlite will ring ,and I answered ,I can hear the voice ,I press >>>> some dtmf,but freeswitch can NOT receive any dtmf. really weird. >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> At 2017-06-12 17:37:49, "Giovanni Maruzzelli" >>>> wrote: >>>> >>>> you need to check the DTMF type, you probably are using the wrong one >>>> (info-inband-rfc2833), and for some reason they are not negotiated >>>> >>>> >>>> On 12 June 2017 at 02:32, chenyzhi wrote: >>>> >>>>> I have read the logs ,but I didn't find any difference. >>>>> >>>>> >>>>> Please make a test to see if this happens in your box. >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> At 2017-06-10 06:49:34, "David Villasmil" < >>>>> david.villasmil.work at gmail.com> wrote: >>>>> >>>>> Have you looked at the log? Bump the logging up and see what shows >>>>> up... what you're seeing is very weird >>>>> >>>>> David >>>>> >>>>> On Fri, Jun 9, 2017 at 11:18 AM chenyzhi wrote: >>>>> >>>>>> Hi >>>>>> >>>>>> >>>>>> call 5000 from x-lite , you can hear the IVR voice and if you press >>>>>> dtmf keys , freeswitch can receive the dtmf keys . >>>>>> >>>>>> >>>>>> but , if you enter the command "originate user/1001 5000" , x-lite >>>>>> will ring ,answer it ,you can hear the IVR voice , press some keys ,the >>>>>> freeswitch can NOT receive any dtmf , >>>>>> >>>>>> >>>>>> why? >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> ____________________________________________________________ >>>>>> _____________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at 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 at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>>>> switch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>> >>>>> >>>>> >>>>> >>>>> ____________________________________________________________ >>>>> _____________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at 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 at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>>> switch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> >>>> -- >>>> >>>> >>>> Sincerely, >>>> >>>> Giovanni Maruzzelli >>>> OpenTelecom.IT >>>> cell: +39 347 266 56 18 >>>> >>>> >>>> >>>> >>>> >>>> ____________________________________________________________ >>>> _____________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at 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 at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>> switch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> >>> >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at 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 at 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 at 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 at 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 > OpenTelecom.IT > cell: +39 347 266 56 18 > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at 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 OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: From gmaruzz at gmail.com Thu Jun 15 09:19:09 2017 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Thu, 15 Jun 2017 11:19:09 +0200 Subject: [Freeswitch-users] Call recording In-Reply-To: <7c47071f86ce436186b4ab8bf58de46c@w12mad02.comunycarse.com> References: <7c47071f86ce436186b4ab8bf58de46c@w12mad02.comunycarse.com> Message-ID: On 15 June 2017 at 09:49, Sven Uszpelkat wrote: > Hello, > > > > We are using FreeSWITCH as a third-party recording application, i.e. we > are receiving SIP calls with the complete audio of conversations taking > place on another switch and we are saving this audio to a file. To achieve > this we are using a simple script similar to this: > > > > session:answer() > > while(session:ready() == true) do > > test = session:recordFile("/usr/local/freeswitch/recordings/test.wav", > 18000, 0, 300) > > session:setAutoHangup(false) > > session:hangup() > > end > > > > This script will be invoked by the following dialplan: > > > > > > > > > > > > > > > > Basically it seems to work quite well, but sometimes there are missing > audio at the end of the recorded file. Usually it’s only a few seconds, > but sometimes it seems to be more. (It’s like the recording sometimes goes > behind the real call and when the hangup event is received the remaining > audio is discarded.) > > > > What could be the reason for this behavior? Is there something wrong with > the script or is there a better way to achieve our goal? > One first question come to my mind: why do you use a script here? A simple extension can do exactly the same, if you just want to record the session... https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools:+record_session Anyway, if you want to use the script, why you first session:setAutoHangup(false) and after that you session:hangup() ? Also, you made the silence_threshold equal 0 (zero). Have you has the same problems using a silence_threshold of, let's say, 30 (thirty), like in documentation? ( https://freeswitch.org/confluence/display/FREESWITCH/Lua+API+Reference#LuaAPIReference-session:recordFile ) Hope this helps, -giovanni > > > Many thanks in advance. > > > > Best regards, > > > > Sven Uszpelkat > Departamento I+D > Comunycarse Network Consultants, S.L. > > > > > > [image: Descripción: Descripción: > http://www.comunycarse.com/email_images/facebook_16.jpg] > [image: > Descripción: Descripción: > http://www.comunycarse.com/email_images/linkedin_16.jpg] > [image: Descripción: > Descripción: http://www.comunycarse.com/email_images/twitter_16.jpg] > [image: Descripción: Descripción: > http://www.comunycarse.com/email_images/wordpress_16.jpg] > > > Joaquín Turina, 2 > 28224 Pozuelo de Alarcón MADRID > Tlf. +34 917 498 700 <+34%20917%2049%2087%2000> > Fax +34 917 498 720 <+34%20917%2049%2087%2020> > > Sabino Arana, 18 > 08028 BARCELONA > Tlf. +34 934 098 480 <+34%20934%2009%2084%2080> > Fax +34 934 098 490 <+34%20934%2009%2084%2090> > > http://www.comunycarse.com > > > AVISO LEGAL > La presente comunicación y sus anexos tiene como destinatario la persona a > la que va dirigida, por lo que si usted lo recibe por error debe > notificarlo al remitente y eliminarlo de su sistema, no pudiendo > utilizarlo, total o parcialmente, para ningún fin. Su contenido puede tener > información confidencial o protegida legalmente y únicamente expresa la > opinión del remitente. El uso del correo electrónico vía internet no > permite asegurar ni la confidencialidad de los mensajes ni su correcta > recepción. En el caso de que el destinatario no consintiera la utilización > del correo electrónico deberá ponerlo en nuestro conocimiento > inmediatamente. > > DISCLAIMER > This message and its attachments are intended exclusively for the named > addressee. If you receive this message by mistake, please delete it > immediately from your system and notify the sender. You may not use this > message or any part of it for any purpose. The message may contain > information that is confidential or protected by law, and any opinions > expressed are those of the individual sender. Internet email guarantees > neither the confidentiality nor the proper receipt of the message sent. If > the addressee of this message does not consent to the use of internet > e-mail, please inform us immediately. > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at 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 OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 1630 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image004.jpg Type: image/jpeg Size: 1402 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image002.jpg Type: image/jpeg Size: 1694 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image003.jpg Type: image/jpeg Size: 1628 bytes Desc: not available URL: From igorolhovskiy at gmail.com Thu Jun 15 09:40:00 2017 From: igorolhovskiy at gmail.com (Igor Olhovskiy) Date: Thu, 15 Jun 2017 12:40:00 +0300 Subject: [Freeswitch-users] question about HA solution In-Reply-To: References: <8A13E0EC-FA13-4267-80F6-CE1A8E8360CF@jerris.com> Message-ID: Hi! Same situation here. Idea is: I’m having Freeswitch HA (keepalived, working, same database, calls recovering…) If I look on «show calls» at slave node, I see calls on master node. I crash master node (with «fsctl crash»), calls are transferred to slave node, restored, but when I run «show calls» on this (slave) node again, I see 0 calls. But calls are actually going on. So, it’s seems impossible to have 2nd recover on already recovered call. In DB logs seen an errors like insert into channels (uuid,direction,created,created_epoch, name,state,callstate,dialplan,context,hostname,initial_cid_name,initial_cid_num,initial_ip_addr,initial_dest,initial_dialplan,initial_context) values('57904410-a8ad-4c28-a88a-83bd2280e146','outbound','2017-06-15 19:30:32','1497519032','sofia/internal/113-akbepcb59gt2a at 172.17.240.50:5060 ','CS_INIT','DOWN','XML','sip303.empowervoice.com ','blueAPACHE_test','103','103','172.17.240.50','113-akbepcb59gt2a','XML',' sip303.empowervoice.com') Jun 15 19:30:31 E2-EVL-T-DB-01 postgres[28042]: [109-1] 2017-06-15 19:30:31 AEST [28042-103] freeswitch at freeswitch ERROR: duplicate key value violates unique constraint «channels_pkey" Or like statement: insert into calls (call_uuid,call_created,call_created_epoch,caller_uuid,callee_uuid,hostname) values ('ffaf3eb5-3fc5-47fe-adef-cc4dddf53bab','2017-06-15 19:30:32','1497519032','ffaf3eb5-3fc5-47fe-adef-cc4dddf53bab','57904410-a8ad-4c28-a88a-83bd2280e146','blueAPACHE_test') Jun 15 19:30:31 E2-EVL-T-DB-01 postgres[28042]: [147-1] 2017-06-15 19:30:31 AEST [28042-142] freeswitch at freeswitch ERROR: duplicate key value violates unique constraint «calls_pkey" Also I see much queries like this delete from calls where (caller_uuid=‘ffaf3eb5-3fc5-47fe-adef-cc4dddf53bab’ or callee_uuid='ffaf3eb5-3fc5-47fe-adef-cc4dddf53bab') delete from recovery where runtime_uuid!=‘91f571c5-e0d2-462e-aa84-e4ca07052119’ and technology=‘sofia’…. when calls are switched. So, can this help to point an issue? 2017-06-08 18:48 GMT+03:00 Michael Jerris : > check your db logs as nothing we are doing should be clearing those. > > On Thu, Jun 8, 2017 at 4:08 AM Denys Pozniak > wrote: > >> Hello! >> >> My configs: >> >> *switch.conf.xml* >> >> >> >> >> >> >> >> >> *external.conf.xml* >> >> >> >> >> >> On 7 June 2017 at 17:35, Michael Jerris wrote: >> >>> That param should keep it from doing so, if its not you are not setting >>> it somehow or something else is wiping the db. >>> >>> On Jun 5, 2017, at 1:50 PM, Denys Pozniak >>> wrote: >>> >>> Yes, correct. But when you restart FS on slave, it will erase database. >>> And option auto-clear-sql=false not working for me. >>> >>> On Jun 5, 2017 6:32 PM, "Michael Jerris" wrote: >>> >>> recovered calls will get new entries in the table. >>> >>> On Jun 5, 2017, at 7:41 AM, Denys Pozniak >>> wrote: >>> >>> Hello! >>> >>> Thank you *Raymond* about your explanation, but I dont agree with some >>> point: >>> *If it really need an answer about your question -- "if it is possible >>> to move calls back". I think it's unnecessary.* - in my case I have >>> two not equal servers, so I need to have only one as a master. >>> If switchover happens I need to have ability to restore master back. >>> >>> Thank you *Luis* for your link, you can do simple test to understand >>> what I am talking about: do call -> check on master and slave #show >>> channels -> restart FS on slave -> check on master #show channels. In my >>> case I dont see any active calls after this, so restoring back is not >>> possible. >>> >>> >>> >>> On 3 June 2017 at 22:16, Luis Daniel Lucio Quiroz < >>> luis.daniel.lucio at gmail.com> wrote: >>> >>>> You may want to read this article. >>>> >>>> http://inside-out.xyz/technology/how-to-configure- >>>> freeswitch-for-ha.html >>>> >>>> Le 31 mai 2017 6:29 PM, "Denys Pozniak" >>>> a écrit : >>>> >>>> Hello! >>>> >>>> I built FS HA solution based on keepalived and mysql master-master. >>>> It works ok generally, but as I understand FS after restarting cleaning >>>> own database. >>>> >>>> So when node1 fails calls jump to node2, after script restarts node1 it >>>> is not possible to move calls back. >>>> >>>> Tried options in switch.conf.xml, but no luck: >>>> >>>> >>>> >>>> >>>> Is there is a way to solve this? >>>> >>>> >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at 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 at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Best regards, Igor -------------- next part -------------- An HTML attachment was scrubbed... URL: From matt at supportedbusiness.com Thu Jun 15 09:54:32 2017 From: matt at supportedbusiness.com (Matt Broad) Date: Thu, 15 Jun 2017 10:54:32 +0100 Subject: [Freeswitch-users] group_confirm_file multiple files Message-ID: Hi, I'm wondering if it is possible to play multiple files using the group_confirm_file function. I have 2 audio files that I would like to play 1 after the other and then wait for the confirm key. I have tried using mod_file_string, but get an error "Error from mpg123: File access error. (code 22)", I assume this is due to the fact it is reading the file string as one file rather than 2 separated by the ! delimiter. thanks Matt Matt Broad Tel: +44 (0)203 011 1313 <+44%2020%203011%201313> Web: www.supportedbusiness.com -------------- next part -------------- An HTML attachment was scrubbed... URL: From s.safarov at gmail.com Thu Jun 15 11:16:35 2017 From: s.safarov at gmail.com (Sergey Safarov) Date: Thu, 15 Jun 2017 11:16:35 +0000 Subject: [Freeswitch-users] Verto failover In-Reply-To: References: Message-ID: Yes. Balancing can be done using: 1) at dns level; 2) using haproxy daemon; 3) using nginx as proxy. чт, 15 июн. 2017 г. в 11:41, Ketan Kothari : > Can we setup Verto communicator *User-Interface* and *FreeSWITCH* on > separate server? > If yes > > *--->* We have 2 FreeSWITCH servers and 1 User-Interface server. So how > to load balance or fail-over of Verto? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From gmaruzz at gmail.com Thu Jun 15 11:16:52 2017 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Thu, 15 Jun 2017 13:16:52 +0200 Subject: [Freeswitch-users] Call recording In-Reply-To: <5323e6cfb2844e59a19957fad4e8044c@w12mad02.comunycarse.com> References: <7c47071f86ce436186b4ab8bf58de46c@w12mad02.comunycarse.com> <5323e6cfb2844e59a19957fad4e8044c@w12mad02.comunycarse.com> Message-ID: On 15 June 2017 at 12:50, Sven Uszpelkat wrote: > 3.) We set the silence threshold to 0 because the documentation is not > very clear how to disable the silence detection. We don’t want the > recording to stop in response to a period of silence. The point is to > record everything. > > The silence_threshold determines what is considered silence, eg below what level of acoustic energy we state the stream is containing silence. Then, we wait for "how_many_silence_seconds" or until hangup before stopping recording. So, maybe you are right, and setting it to 0 will consider silence only when there is absolute silence in the stream, so for all practical purposes, until hangup. I have no mean to check source code now. On another hand, I can think at other possible causes for the premature end of the recorded file: maybe you move the file before it has been flushed by FreeSWITCH or by operating system? Maybe the hangup in the script interrupts the recording in the script and close the file descriptor before is flushed? (I am shooting in the dark) You can try to leave out those two lines, and test again. Also, you can insert a line that sync (flush) the filesystem before exiting, just to be sure. I would insert it after the while(session:ready()) A system(sync), or something similar will probably do. Hope this helps, -giovanni > > > Best regards, > > Sven > > > > *De:* FreeSWITCH-users [mailto:freeswitch-users-bounc > es at lists.freeswitch.org ] *En > nombre de *Giovanni Maruzzelli > *Enviado el:* jueves, 15 de junio de 2017 11:19 > *Para:* FreeSWITCH Users Help > *Asunto:* Re: [Freeswitch-users] Call recording > > > > > > > > On 15 June 2017 at 09:49, Sven Uszpelkat wrote: > > Hello, > > > > We are using FreeSWITCH as a third-party recording application, i.e. we > are receiving SIP calls with the complete audio of conversations taking > place on another switch and we are saving this audio to a file. To achieve > this we are using a simple script similar to this: > > > > session:answer() > > while(session:ready() == true) do > > test = session:recordFile("/usr/local/freeswitch/recordings/test.wav", > 18000, 0, 300) > > session:setAutoHangup(false) > > session:hangup() > > end > > > > This script will be invoked by the following dialplan: > > > > > > > > > > > > > > > > Basically it seems to work quite well, but sometimes there are missing > audio at the end of the recorded file. Usually it’s only a few seconds, > but sometimes it seems to be more. (It’s like the recording sometimes goes > behind the real call and when the hangup event is received the remaining > audio is discarded.) > > > > What could be the reason for this behavior? Is there something wrong with > the script or is there a better way to achieve our goal? > > > > One first question come to my mind: why do you use a script here? A simple > extension can do exactly the same, if you just want to record the session... > > https://freeswitch.org/confluence/display/FREESWITCH/mod_ > dptools:+record_session > > Anyway, if you want to use the script, why you first > > session:setAutoHangup(false) > > and after that you > > > > session:hangup() > > > > ? > > > > Also, you made the silence_threshold equal 0 (zero). > Have you has the same problems using a silence_threshold of, let's say, 30 > (thirty), like in documentation? ( https://freeswitch.org/conflue > nce/display/FREESWITCH/Lua+API+Reference#LuaAPIReference- > session:recordFile ) > > > > Hope this helps, > > -giovanni > > > > > > > Many thanks in advance. > > > > Best regards, > > > > Sven Uszpelkat > Departamento I+D > Comunycarse Network Consultants, S.L. > > > > > > [image: Descripción: Descripción: > http://www.comunycarse.com/email_images/facebook_16.jpg] > [image: > Descripción: Descripción: > http://www.comunycarse.com/email_images/linkedin_16.jpg] > [image: Descripción: > Descripción: http://www.comunycarse.com/email_images/twitter_16.jpg] > [image: Descripción: Descripción: > http://www.comunycarse.com/email_images/wordpress_16.jpg] > > > Joaquín Turina, 2 > 28224 Pozuelo de Alarcón MADRID > Tlf. +34 917 498 700 <+34%20917%2049%2087%2000> > Fax +34 917 498 720 <+34%20917%2049%2087%2020> > > Sabino Arana, 18 > 08028 BARCELONA > Tlf. +34 934 098 480 <+34%20934%2009%2084%2080> > Fax +34 934 098 490 <+34%20934%2009%2084%2090> > > http://www.comunycarse.com > > > AVISO LEGAL > La presente comunicación y sus anexos tiene como destinatario la persona a > la que va dirigida, por lo que si usted lo recibe por error debe > notificarlo al remitente y eliminarlo de su sistema, no pudiendo > utilizarlo, total o parcialmente, para ningún fin. Su contenido puede tener > información confidencial o protegida legalmente y únicamente expresa la > opinión del remitente. El uso del correo electrónico vía internet no > permite asegurar ni la confidencialidad de los mensajes ni su correcta > recepción. En el caso de que el destinatario no consintiera la utilización > del correo electrónico deberá ponerlo en nuestro conocimiento > inmediatamente. > > DISCLAIMER > This message and its attachments are intended exclusively for the named > addressee. If you receive this message by mistake, please delete it > immediately from your system and notify the sender. You may not use this > message or any part of it for any purpose. The message may contain > information that is confidential or protected by law, and any opinions > expressed are those of the individual sender. Internet email guarantees > neither the confidentiality nor the proper receipt of the message sent. If > the addressee of this message does not consent to the use of internet > e-mail, please inform us immediately. > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at 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 > OpenTelecom.IT > cell: +39 347 266 56 18 > -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image003.jpg Type: image/jpeg Size: 1628 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image002.jpg Type: image/jpeg Size: 1694 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 1630 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image004.jpg Type: image/jpeg Size: 1402 bytes Desc: not available URL: From adrian.worutowicz at esifrance.net Thu Jun 15 09:44:16 2017 From: adrian.worutowicz at esifrance.net (Adrian Worutowicz) Date: Thu, 15 Jun 2017 11:44:16 +0200 Subject: [Freeswitch-users] Build Problem in VS2015 Message-ID: <001001d2e5bb$f4519cc0$dcf4d640$@worutowicz@esifrance.net> Hello, I try to recompile FS in VS2015 without success. I took FS sources from git master. git config --global core.autocrlf false git clone https://stash.freeswitch.org/scm/fs/freeswitch.git /c/ESI/Components/FreeSwitch/ I have wix311 for VS2015 installed. For example it searches in folder 'openssl-1.0.2k' while only a folder 'openssl' exists. I tried to recompile mod_PortAudio, and I got c:\ESI\components\freeswitch\src\mod\endpoints\mod_portaudio\pablio.h(55): fatal error C1083: Impossible d'ouvrir le fichier include : 'portaudio.h' : No such file or directory. Indeed 'portaudio.h' does not exist. Plenty of other errors in the attached file. What do I miss? Thanks in advance, Adrian. -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: vs2015BuildReport.zip Type: application/octet-stream Size: 13747 bytes Desc: not available URL: From USZPELSV at comunycarse.com Thu Jun 15 10:50:03 2017 From: USZPELSV at comunycarse.com (Sven Uszpelkat) Date: Thu, 15 Jun 2017 10:50:03 +0000 Subject: [Freeswitch-users] Call recording In-Reply-To: References: <7c47071f86ce436186b4ab8bf58de46c@w12mad02.comunycarse.com> Message-ID: <5323e6cfb2844e59a19957fad4e8044c@w12mad02.comunycarse.com> Hi Giovanni, Thank you for your response. 1.) The code example I provided was only a fragment of the script. The actual script is doing more than just recording but we will try the dialplan approach as well. 2.) The session:hangup was thought to leave the while loop but maybe it’s not necessary if the session is already finished or we change the while for an if. I think I copied this from another example I found. 3.) We set the silence threshold to 0 because the documentation is not very clear how to disable the silence detection. We don’t want the recording to stop in response to a period of silence. The point is to record everything. Best regards, Sven De: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] En nombre de Giovanni Maruzzelli Enviado el: jueves, 15 de junio de 2017 11:19 Para: FreeSWITCH Users Help Asunto: Re: [Freeswitch-users] Call recording On 15 June 2017 at 09:49, Sven Uszpelkat > wrote: Hello, We are using FreeSWITCH as a third-party recording application, i.e. we are receiving SIP calls with the complete audio of conversations taking place on another switch and we are saving this audio to a file. To achieve this we are using a simple script similar to this: session:answer() while(session:ready() == true) do test = session:recordFile("/usr/local/freeswitch/recordings/test.wav", 18000, 0, 300) session:setAutoHangup(false) session:hangup() end This script will be invoked by the following dialplan: Basically it seems to work quite well, but sometimes there are missing audio at the end of the recorded file. Usually it’s only a few seconds, but sometimes it seems to be more. (It’s like the recording sometimes goes behind the real call and when the hangup event is received the remaining audio is discarded.) What could be the reason for this behavior? Is there something wrong with the script or is there a better way to achieve our goal? One first question come to my mind: why do you use a script here? A simple extension can do exactly the same, if you just want to record the session... https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools:+record_session Anyway, if you want to use the script, why you first session:setAutoHangup(false) and after that you session:hangup() ? Also, you made the silence_threshold equal 0 (zero). Have you has the same problems using a silence_threshold of, let's say, 30 (thirty), like in documentation? ( https://freeswitch.org/confluence/display/FREESWITCH/Lua+API+Reference#LuaAPIReference-session:recordFile ) Hope this helps, -giovanni Many thanks in advance. Best regards, Sven Uszpelkat Departamento I+D Comunycarse Network Consultants, S.L. [Descripción: Descripción: http://www.comunycarse.com/email_images/facebook_16.jpg] [Descripción: Descripción: http://www.comunycarse.com/email_images/linkedin_16.jpg] [Descripción: Descripción: http://www.comunycarse.com/email_images/twitter_16.jpg] [Descripción: Descripción: http://www.comunycarse.com/email_images/wordpress_16.jpg] Joaquín Turina, 2 28224 Pozuelo de Alarcón MADRID Tlf. +34 917 498 700 Fax +34 917 498 720 Sabino Arana, 18 08028 BARCELONA Tlf. +34 934 098 480 Fax +34 934 098 490 http://www.comunycarse.com AVISO LEGAL La presente comunicación y sus anexos tiene como destinatario la persona a la que va dirigida, por lo que si usted lo recibe por error debe notificarlo al remitente y eliminarlo de su sistema, no pudiendo utilizarlo, total o parcialmente, para ningún fin. Su contenido puede tener información confidencial o protegida legalmente y únicamente expresa la opinión del remitente. El uso del correo electrónico vía internet no permite asegurar ni la confidencialidad de los mensajes ni su correcta recepción. En el caso de que el destinatario no consintiera la utilización del correo electrónico deberá ponerlo en nuestro conocimiento inmediatamente. DISCLAIMER This message and its attachments are intended exclusively for the named addressee. If you receive this message by mistake, please delete it immediately from your system and notify the sender. You may not use this message or any part of it for any purpose. The message may contain information that is confidential or protected by law, and any opinions expressed are those of the individual sender. Internet email guarantees neither the confidentiality nor the proper receipt of the message sent. If the addressee of this message does not consent to the use of internet e-mail, please inform us immediately. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at 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 at 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 OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 1630 bytes Desc: image001.jpg URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image002.jpg Type: image/jpeg Size: 1694 bytes Desc: image002.jpg URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image003.jpg Type: image/jpeg Size: 1628 bytes Desc: image003.jpg URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image004.jpg Type: image/jpeg Size: 1402 bytes Desc: image004.jpg URL: From kkothari157 at gmail.com Thu Jun 15 12:55:32 2017 From: kkothari157 at gmail.com (Ketan Kothari) Date: Thu, 15 Jun 2017 18:25:32 +0530 Subject: [Freeswitch-users] Verto failover In-Reply-To: References: Message-ID: Hello Sergey, Thanks for your response. *3) using nginx as proxy.* Could you please tell me some reference to do it. On Thu, Jun 15, 2017 at 4:46 PM, Sergey Safarov wrote: > Yes. > Balancing can be done using: > 1) at dns level; > 2) using haproxy daemon; > 3) using nginx as proxy. > > чт, 15 июн. 2017 г. в 11:41, Ketan Kothari : > >> Can we setup Verto communicator *User-Interface* and *FreeSWITCH* on >> separate server? >> If yes >> >> *--->* We have 2 FreeSWITCH servers and 1 User-Interface server. So how >> to load balance or fail-over of Verto? >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From rmundkowsky at ets.org Thu Jun 15 13:51:53 2017 From: rmundkowsky at ets.org (Mundkowsky, Robert) Date: Thu, 15 Jun 2017 13:51:53 +0000 Subject: [Freeswitch-users] Verto failover In-Reply-To: References: Message-ID: Verto is just frontend client side code (javascript), you can put it anywhere and point to a FreeSWITCH server anywhere else. I am curious what type of failover is handled by Sergey’s suggestion. Would this support an active call to continue? Or are you talking about failover for next call? Robert Mundkowsky From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sergey Safarov Sent: Thursday, June 15, 2017 7:17 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Verto failover Yes. Balancing can be done using: 1) at dns level; 2) using haproxy daemon; 3) using nginx as proxy. чт, 15 июн. 2017 г. в 11:41, Ketan Kothari >: Can we setup Verto communicator User-Interface and FreeSWITCH on separate server? If yes ---> We have 2 FreeSWITCH servers and 1 User-Interface server. So how to load balance or fail-over of Verto? _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ________________________________ This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited. Thank you for your compliance. ________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL: From soapee01.fs at stubbornroses.com Thu Jun 15 22:06:14 2017 From: soapee01.fs at stubbornroses.com (soapee01.fs at stubbornroses.com) Date: Thu, 15 Jun 2017 17:06:14 -0500 Subject: [Freeswitch-users] Missed Calls with Hunt Group Message-ID: <594304D6.7070207@stubbornroses.com> Hi, Here's the command that I'm running. originate user/102 at domain &bridge(user/100 at domain,user/101 at domain) On FS version 1.2.22, if the user 100 answers the call, user 101 will not see a missed call notification. On FS version FS 1.6.5 if the user 100 answers the call, user 101 will show a missed call. Is there something in the docs I've missed? I'd really like to set it back to the old way, but it would be really cool if there's a variable somewhere I've missed that lets you choose the behavior. Thanks! James From jungleboogie0 at gmail.com Thu Jun 15 23:30:45 2017 From: jungleboogie0 at gmail.com (jungle Boogie) Date: Thu, 15 Jun 2017 16:30:45 -0700 Subject: [Freeswitch-users] =?utf-8?q?FreeSWITCH_Week_in_Review_=28Master_?= =?utf-8?q?Branch=29_December_5th_=E2=80=93_December_12th?= In-Reply-To: <5671969d94b11_2c9b52d334780b@resque-worker.8.mail> References: <5671969d94b11_2c9b52d334780b@resque-worker.8.mail> Message-ID: Hi All, On 16 December 2015 at 08:51, Ken Rice wrote: > New Post on freeswitch.org from Kathleen King > check it out at http://ift.tt/1P8dyo3 > FreeSWITCH Week in Review (Master Branch) December 5th – December 12th What happened with these emails? They're really informative and give people an idea on what's changing. From brian at freeswitch.org Fri Jun 16 00:23:30 2017 From: brian at freeswitch.org (Brian West) Date: Thu, 15 Jun 2017 19:23:30 -0500 Subject: [Freeswitch-users] Missed Calls with Hunt Group In-Reply-To: <594304D6.7070207@stubbornroses.com> References: <594304D6.7070207@stubbornroses.com> Message-ID: Verify the answered elsewhere header is there, Thats what makes that happen, plus try 1.6.18. /b On Thu, Jun 15, 2017 at 5:06 PM, wrote: > Hi, > > Here's the command that I'm running. > > originate user/102 at domain &bridge(user/100 at domain,user/101 at domain) > > > On FS version 1.2.22, if the user 100 answers the call, user 101 will not > see a missed call notification. > > On FS version FS 1.6.5 if the user 100 answers the call, user 101 will > show a missed call. > > Is there something in the docs I've missed? I'd really like to set it back > to the old way, but it would be really cool if there's a variable somewhere > I've missed that lets you choose the behavior. > > Thanks! > > James > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Book a phone call (CST) Allison prompts for FreeSWITCH: *https://www.gofundme.com/allison-prompts-for-freeswitch* Got Bugs? Report them here ! | Reddit: /r/freeswitch *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: From tculjaga at gmail.com Fri Jun 16 08:57:14 2017 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Fri, 16 Jun 2017 10:57:14 +0200 Subject: [Freeswitch-users] WebRTC and Carrier NAT ( CgNAT) Message-ID: hello, does anyone have experience with WebRTC via carrier NAT ? client_LAN(192.168.x.y) carrer_prvate_WAN(10.x.x.x) carrier_public_WAN(public_ip) <> internet in this scenario, WebRTC calls using STUN only will not work. What about TURN ? did anyone try that ? Regards, Tihomir. -------------- next part -------------- An HTML attachment was scrubbed... URL: From hardikitpl at gmail.com Fri Jun 16 09:23:20 2017 From: hardikitpl at gmail.com (Hardik Patel) Date: Fri, 16 Jun 2017 14:53:20 +0530 Subject: [Freeswitch-users] Fax receive issue with t30 codec In-Reply-To: References: Message-ID: Hi All Finally.......!!! It works for me and it's nice experience to troubleshoot on it. Thank you very much you all who support me on this. Thanks a lot. On Wed, Jun 14, 2017 at 5:28 PM, Giovanni Maruzzelli wrote: > try setting the transfer to 9600 (fax_disable_v17) > > Also, you may find this page useful: > > https://freeswitch.org/confluence/display/FREESWITCH/ > mod_spandsp#mod_spandsp-Fax > > > > On 14 June 2017 at 13:52, Hardik Patel wrote: > >> Hi Brian, >> >> Is there any solution for that? >> >> On Sat, Jun 10, 2017 at 9:46 AM, Hardik Patel >> wrote: >> >>> Hi Brian, >>> >>> Thanks for the support. >>> >>> We are testing receive fax functionality using real fax machine and here >>> i have listed the model which we are using to send fax. >>> >>> the models of real fax machines that we have used are group 3 CCITT / >>> ITU, they are the following: >>> 1 ) konica minolta bizhub-c220 >>> 2 ) HP Officejet 4500 >>> 3 ) HP Officejet G85 >>> >>> >>> From above list one of our machine is sending fax without T38 support >>> and we got the failure with same error which have posted on bug but if we >>> use T38 support then it works fine for us. >>> >>> >>> >>> *ERROR :* >>> 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:508 Fax processing >>> not successful - result (3) Timed out waiting for the first message. >>> 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:513 Remote station >>> id: >>> 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:514 Local station >>> id: SpanDSP Fax Ident >>> 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:515 Pages >>> transferred: 0 >>> 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:517 Total fax >>> pages: 0 >>> 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:518 Image >>> resolution: 0x0 >>> 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:519 Transfer >>> Rate: 14400 >>> 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:521 ECM >>> status off >>> 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:522 remote >>> country: >>> 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:523 remote >>> vendor: >>> 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:524 remote >>> model: >>> 2017-06-09 11:48:16.537520 [DEBUG] mod_spandsp_fax.c:526 >>> ============================== >>> >>> On Sat, Jun 10, 2017 at 2:42 AM, Brian West >>> wrote: >>> >>>> Your log snip doesn't really help, I know without a single doubt faxing >>>> works fine. So what are you doing and how are you doing it? >>>> >>>> On Fri, Jun 9, 2017 at 3:42 PM, Tihomir Culjaga >>>> wrote: >>>> >>>>> hi, whats new on faxing in 1.6.17 ? >>>>> >>>>> T. >>>>> >>>>> On 6 June 2017 at 16:00, Brian West wrote: >>>>> >>>>>> You'll have to use 1.6.17 if you ever want any faxing to work in all >>>>>> test cases. >>>>>> >>>>>> /b >>>>>> >>>>>> On Tue, Jun 6, 2017 at 6:11 AM, Hardik Patel >>>>>> wrote: >>>>>> >>>>>>> Hello, >>>>>>> >>>>>>> I am using opensips as entry point using dispatcher. opensips( >>>>>>> 127.0.0.1), i am routing call to freeswitch server (127.0.0.3). >>>>>>> >>>>>>> Now I am trying to receive fax, my issue is when i try to send fax >>>>>>> in softphone(Zoiper) from the log i am seeing that it is sending fax using >>>>>>> t30 codec. and i am not receiving the fax at destination, is it because of >>>>>>> codec, should it only work with t38 codec? if that is the issue than how am >>>>>>> i be able to send the fax using t38 from zoiper? >>>>>>> >>>>>>> Here i am attaching the fs log with loglevel 9 and sip trace is also >>>>>>> enabled. >>>>>>> >>>>>>> 127.0.0.2 => carrier/provider IP >>>>>>> 123456789 => Fax number >>>>>>> test at gamil.com => Email Address >>>>>>> 127.0.0.4 =>UI IP >>>>>>> >>>>>>> Pastebin link:https://pastebin.freeswitch.org/view/9ec52715 >>>>>>> >>>>>>> ____________________________________________________________ >>>>>>> _____________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at 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 at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>>>>> switch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> >>>>>> *Brian West* >>>>>> brian at freeswitch.org >>>>>> >>>>>> *Twitter: @FreeSWITCH , @briankwest* >>>>>> >>>>>> http://www.freeswitchbook.com >>>>>> http://www.freeswitchcookbook.com >>>>>> >>>>>> Book a phone call (CST) >>>>>> >>>>>> Allison prompts for FreeSWITCH: >>>>>> >>>>>> *https://www.gofundme.com/allison-prompts-for-freeswitch* >>>>>> >>>>>> >>>>>> Got Bugs? Report them here ! | Reddit: >>>>>> /r/freeswitch >>>>>> >>>>>> *T:*+19184209001 <+1%20918-420-9001> | *F:*+19184209002 >>>>>> <+1%20918-420-9002> | *M:*+1918424WEST (9378) >>>>>> *Skype:*briankwest >>>>>> >>>>>> ____________________________________________________________ >>>>>> _____________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at 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 at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>>>> switch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>> >>>>> >>>>> ____________________________________________________________ >>>>> _____________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at 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 at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>>> switch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> >>>> -- >>>> >>>> *Brian West* >>>> brian at freeswitch.org >>>> >>>> *Twitter: @FreeSWITCH , @briankwest* >>>> >>>> http://www.freeswitchbook.com >>>> http://www.freeswitchcookbook.com >>>> >>>> Book a phone call (CST) >>>> >>>> Allison prompts for FreeSWITCH: >>>> >>>> *https://www.gofundme.com/allison-prompts-for-freeswitch* >>>> >>>> >>>> Got Bugs? Report them here ! | Reddit: >>>> /r/freeswitch >>>> >>>> *T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002 >>>> <(918)%20420-9002> | *M:*+1918424WEST (9378) >>>> *Skype:*briankwest >>>> >>>> ____________________________________________________________ >>>> _____________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at 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 at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>> switch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> >>> -- >>> Hardik Patel >>> iNextrix Technologies Pvt Ltd >>> >> >> >> >> -- >> Hardik Patel >> iNextrix Technologies Pvt Ltd >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at 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 > OpenTelecom.IT > cell: +39 347 266 56 18 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Hardik Patel iNextrix Technologies Pvt Ltd -------------- next part -------------- An HTML attachment was scrubbed... URL: From kkothari157 at gmail.com Fri Jun 16 07:02:39 2017 From: kkothari157 at gmail.com (Ketan Kothari) Date: Fri, 16 Jun 2017 12:32:39 +0530 Subject: [Freeswitch-users] Verto failover In-Reply-To: References: Message-ID: Hello Robert, I'm looking for fail-over for next call and web-interface as well as if one FreeSWITCH failed then verto continue working from other FreeSWITCH server. If you have any suggestion or reference link then please pass it will helpful for me. On Thu, Jun 15, 2017 at 7:21 PM, Mundkowsky, Robert wrote: > Verto is just frontend client side code (javascript), you can put it > anywhere and point to a FreeSWITCH server anywhere else. > > > > I am curious what type of failover is handled by Sergey’s suggestion. > > > > Would this support an active call to continue? Or are you talking about > failover for next call? > > > > > > Robert Mundkowsky > > > > *From:* FreeSWITCH-users [mailto:freeswitch-users- > bounces at lists.freeswitch.org] *On Behalf Of *Sergey Safarov > *Sent:* Thursday, June 15, 2017 7:17 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Verto failover > > > > Yes. > Balancing can be done using: > > 1) at dns level; > > 2) using haproxy daemon; > > 3) using nginx as proxy. > > > > чт, 15 июн. 2017 г. в 11:41, Ketan Kothari : > > Can we setup Verto communicator *User-Interface* and *FreeSWITCH* on > separate server? > If yes > > *--->* We have 2 FreeSWITCH servers and 1 User-Interface server. So how > to load balance or fail-over of Verto? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > ------------------------------ > > This e-mail and any files transmitted with it may contain privileged or > confidential information. It is solely for use by the individual for whom > it is intended, even if addressed incorrectly. If you received this e-mail > in error, please notify the sender; do not disclose, copy, distribute, or > take any action in reliance on the contents of this information; and delete > it from your system. Any other use of this e-mail is prohibited. > > Thank you for your compliance. > ------------------------------ > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From eastour at 163.com Fri Jun 16 04:02:14 2017 From: eastour at 163.com (chenyzhi) Date: Fri, 16 Jun 2017 12:02:14 +0800 (CST) Subject: [Freeswitch-users] When calling out , freeswitch doesn't get DTMF ? In-Reply-To: References: <15c898b5d58.2779.b07ebdf329620b8089087c7205b03f01@xbipin.com> <310cf71c.8fc3.15c8ba9c4f6.Coremail.eastour@163.com> <2a18422f.ef8.15c99b7b9bb.Coremail.eastour@163.com> <6c11dd0f.c2c1.15c9c727dc7.Coremail.eastour@163.com> <2f48505d.d9c5.15ca1d28424.Coremail.eastour@163.com> <59401B92020000310000A34D@mail.tedssupply.com> <59412DB4020000310000A381@mail.tedssupply.com> <7d1e3f12.4750.15ca994d89d.Coremail.eastour@163.com> Message-ID: <7611bb49.72be.15caf10d2a0.Coremail.eastour@163.com> Yes ,I can hear all the IVR prompt voices correctly. I don't think it's a NAT problem ,because both the x-lite and the freeswitch are in the same LAN. The sip trace log is in the attatchment. Thank you. PS I tested this on another freeswitch box ,version: FreeSWITCH Version 1.6.16+git~20170403T142423Z~e6d643b29c~32bit (git e6d643b 2017-04-03 14:24:23Z 32bit) It can detect dtmf on outgoing calls. Maybe this only happens on FreeSWITCH Version 1.9.0+git~20170518T231917Z~a1fc18aee5~32bit (git a1fc18a 2017-05-18 23:19:17Z 32bit) 在 2017-06-15 17:04:48，"Giovanni Maruzzelli" 写道： On 15 June 2017 at 04:28, chenyzhi wrote: the version is: FreeSWITCH Version 1.9.0+git~20170518T231917Z~a1fc18aee5~32bit (git a1fc18a 2017-05-18 23:19:17Z 32bit) the complete, since beginning to end, unedited, debug output of console when making a outgoing call which does not get DTMFs and the whole conf folder is in the attatchment. from the log, seems it does not read all the IVR messages correctly to you, it exits straight away... are you able to correctly hear all the IVR messages? also, can you take a SIP trace? (from console: "sofia global siptrace on") I suspect you have a NAT problem of some sort Also, I see you are on MASTER git, on Windows, and on 32 bit... Not sure this is supported... Have you has this problems with stable branch (1.6.x)? thx! 在 2017-06-15 00:42:11，"Giovanni Maruzzelli" 写道： On 14 June 2017 at 18:36, admin wrote: I don't mean to hijack the OP concern with my problem, I just wanted to reinforce I have seen this problem. And the problem is that the call originated through FS appears not to send user entered DTMF to the receiving phone. Let's help chenyzhi and then I'll take a turn. I just tested right now from console with: bgapi originate user/1011 5000 and originate user/1011 5000 and it works Please pastebin your FreeSWITCH version (eg, type "version" in console), your dialplan, your SIP profile, and the complete, since beginning to end, unedited, debug output of console when receiving a call which does not get DTMFs Maybe helping you we'll help him too - James >>> Giovanni Maruzzelli 6/14/2017 03:53 AM >>> Never heard such problems Please pastebin your dialplan, your SIP profile, and the complete, since beginning to end, unedited, debug output of console when receiving a call which does not get DTMFs sent from mobile cell: +39 347 266 56 18 Giovanni Maruzzelli OpenTelecom.IT On Jun 13, 2017 23:07, "admin" wrote: I am encountering the same issue. I am using an ESL call to api originate, with little else except getting in and out of the port, and the caller DTMF is failing to reach the called number. The called number auto answer attendant does not respond to DTMF, and a call to a test phone confirms no DTMF. This is a new upgrade from 1.2 to 1.6 and I don't know if this was an issue in 1.2, but my users never complained before 1.6 upgrade. Ideas?... >>> chenyzhi 06/13/17 3:32 PM >>> It's a one leg call .there is no b-leg. please make a test on your freeswitch box . just type the command "originate user/1001 5000" on the freeswitch console to see if your freeswitch instance can detect dtmf input. At 2017-06-12 22:57:17, "Luis Daniel Lucio Quiroz" wrote: check if you have transcoding, and if you do, check that dftm type-codec on leg b are compatible. -- Luis Daniel Lucio Quiroz CISSP, CISM, CISA Linux, VoIP and much more fun www.okay.com.mx Need LCR? Check out LCR for FusionPBX with FreeSWITCH Need Billing? Check out Billing for FusionPBX with FreeSWITCH On Mon, Jun 12, 2017 at 9:16 AM, chenyzhi wrote: I don't think It's the DTMF type , because when I dial 5000 from x-lite (which has registered to freeswitch as 1001) ,I can hear the voice and if I press any dtmf on x-lite, freeswitch can recieve the dtmfs. This means that the DTMF type is correct ,otherwise freeswitch coudn't have received the dtmfs; when I enter the command "originate user/1001 5000" at the freeswitch console , my xlite will ring ,and I answered ,I can hear the voice ,I press some dtmf,but freeswitch can NOT receive any dtmf. really weird. At 2017-06-12 17:37:49, "Giovanni Maruzzelli" wrote: you need to check the DTMF type, you probably are using the wrong one (info-inband-rfc2833), and for some reason they are not negotiated On 12 June 2017 at 02:32, chenyzhi wrote: I have read the logs ,but I didn't find any difference. Please make a test to see if this happens in your box. At 2017-06-10 06:49:34, "David Villasmil" wrote: Have you looked at the log? Bump the logging up and see what shows up... what you're seeing is very weird David On Fri, Jun 9, 2017 at 11:18 AM chenyzhi wrote: Hi call 5000 from x-lite , you can hear the IVR voice and if you press dtmf keys , freeswitch can receive the dtmf keys . but , if you enter the command "originate user/1001 5000" , x-lite will ring ,answer it ,you can hear the IVR voice , press some keys ,the freeswitch can NOT receive any dtmf , why? _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at 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 at 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 at 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 at 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 OpenTelecom.IT cell: +39 347 266 56 18 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at 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 at 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 at 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 at 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 at 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 at 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 OpenTelecom.IT cell: +39 347 266 56 18 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at 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 at 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 OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: sip-trace-on-log-calling-out-not-receiving-dtmf-log.txt URL: From krice at tollfreegateway.com Thu Jun 15 16:28:59 2017 From: krice at tollfreegateway.com (krice at tollfreegateway.com) Date: Thu, 15 Jun 2017 11:28:59 -0500 Subject: [Freeswitch-users] Build Problem in VS2015 In-Reply-To: <001001d2e5bb$f4519cc0$dcf4d640$@worutowicz@esifrance.net> References: <001001d2e5bb$f4519cc0$dcf4d640$@worutowicz@esifrance.net> Message-ID: <80c901d2e5f4$7e26f930$7a74eb90$@tollfreegateway.com> Not sure whats you are doing incorrect here, but I have just built master, I use the built in git bits with VS2015, and then drop to a command prompt (via the team explorer tab, select branches, right click the repo and select open command prompt) Then git pull, git clean -fdx, git reset -hard origin/master , git pull >From here back to the solution explorer open the FreeSWITCH.2015 solution file and build as normal. I think you have something skewed there old ssl vs new ssl bits From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Adrian Worutowicz Sent: Thursday, June 15, 2017 4:44 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Build Problem in VS2015 Hello, I try to recompile FS in VS2015 without success. I took FS sources from git master. git config --global core.autocrlf false git clone https://stash.freeswitch.org/scm/fs/freeswitch.git /c/ESI/Components/FreeSwitch/ I have wix311 for VS2015 installed. For example it searches in folder 'openssl-1.0.2k' while only a folder 'openssl' exists. I tried to recompile mod_PortAudio, and I got c:\ESI\components\freeswitch\src\mod\endpoints\mod_portaudio\pablio.h(55): fatal error C1083: Impossible d'ouvrir le fichier include : 'portaudio.h' : No such file or directory. Indeed 'portaudio.h' does not exist. Plenty of other errors in the attached file. What do I miss? Thanks in advance, Adrian. -------------- next part -------------- An HTML attachment was scrubbed... URL: From USZPELSV at comunycarse.com Fri Jun 16 07:53:30 2017 From: USZPELSV at comunycarse.com (Sven Uszpelkat) Date: Fri, 16 Jun 2017 07:53:30 +0000 Subject: [Freeswitch-users] Call recording In-Reply-To: References: <7c47071f86ce436186b4ab8bf58de46c@w12mad02.comunycarse.com> <5323e6cfb2844e59a19957fad4e8044c@w12mad02.comunycarse.com> Message-ID: Hi Giovanni, Thank you for your help. We changed our script to something like this: session:answer() while(session:ready() == true) do test = session:recordFile("/usr/local/freeswitch/recordings/test.wav", 18000, 0, 300) os.execute("sync"); session:setAutoHangup(false) session:hangup() end As result we note an improvement primarily in short recordings (<1:30 min). These are now practically all complete. In longer recordings there are still losses and it seems that they are increasing with the duration of the recording. I’m not sure how to interpret this but to me it looks like this: with the sync call we achieved to write the buffer content to the file, however in longer calls there is remaining audio which hasn’t even been read to the buffer. Is that correct? If so then the recording function doesn’t ensure to read the remaining audio after hangup. (With tcdump we checked that all audio packets arrived before the BYE message) Does this mean that this behavior is by design? Best regards, Sven De: Giovanni Maruzzelli [mailto:gmaruzz at gmail.com] Enviado el: jueves, 15 de junio de 2017 13:17 Para: Sven Uszpelkat CC: FreeSWITCH Users Help Asunto: Re: [Freeswitch-users] Call recording On 15 June 2017 at 12:50, Sven Uszpelkat > wrote: 3.) We set the silence threshold to 0 because the documentation is not very clear how to disable the silence detection. We don’t want the recording to stop in response to a period of silence. The point is to record everything. The silence_threshold determines what is considered silence, eg below what level of acoustic energy we state the stream is containing silence. Then, we wait for "how_many_silence_seconds" or until hangup before stopping recording. So, maybe you are right, and setting it to 0 will consider silence only when there is absolute silence in the stream, so for all practical purposes, until hangup. I have no mean to check source code now. On another hand, I can think at other possible causes for the premature end of the recorded file: maybe you move the file before it has been flushed by FreeSWITCH or by operating system? Maybe the hangup in the script interrupts the recording in the script and close the file descriptor before is flushed? (I am shooting in the dark) You can try to leave out those two lines, and test again. Also, you can insert a line that sync (flush) the filesystem before exiting, just to be sure. I would insert it after the while(session:ready()) A system(sync), or something similar will probably do. Hope this helps, -giovanni Best regards, Sven De: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] En nombre de Giovanni Maruzzelli Enviado el: jueves, 15 de junio de 2017 11:19 Para: FreeSWITCH Users Help Asunto: Re: [Freeswitch-users] Call recording On 15 June 2017 at 09:49, Sven Uszpelkat > wrote: Hello, We are using FreeSWITCH as a third-party recording application, i.e. we are receiving SIP calls with the complete audio of conversations taking place on another switch and we are saving this audio to a file. To achieve this we are using a simple script similar to this: session:answer() while(session:ready() == true) do test = session:recordFile("/usr/local/freeswitch/recordings/test.wav", 18000, 0, 300) session:setAutoHangup(false) session:hangup() end This script will be invoked by the following dialplan: Basically it seems to work quite well, but sometimes there are missing audio at the end of the recorded file. Usually it’s only a few seconds, but sometimes it seems to be more. (It’s like the recording sometimes goes behind the real call and when the hangup event is received the remaining audio is discarded.) What could be the reason for this behavior? Is there something wrong with the script or is there a better way to achieve our goal? One first question come to my mind: why do you use a script here? A simple extension can do exactly the same, if you just want to record the session... https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools:+record_session Anyway, if you want to use the script, why you first session:setAutoHangup(false) and after that you session:hangup() ? Also, you made the silence_threshold equal 0 (zero). Have you has the same problems using a silence_threshold of, let's say, 30 (thirty), like in documentation? ( https://freeswitch.org/confluence/display/FREESWITCH/Lua+API+Reference#LuaAPIReference-session:recordFile ) Hope this helps, -giovanni Many thanks in advance. Best regards, Sven Uszpelkat Departamento I+D Comunycarse Network Consultants, S.L. [Descripción: Descripción: http://www.comunycarse.com/email_images/facebook_16.jpg] [Descripción: Descripción: http://www.comunycarse.com/email_images/linkedin_16.jpg] [Descripción: Descripción: http://www.comunycarse.com/email_images/twitter_16.jpg] [Descripción: Descripción: http://www.comunycarse.com/email_images/wordpress_16.jpg] Joaquín Turina, 2 28224 Pozuelo de Alarcón MADRID Tlf. +34 917 498 700 Fax +34 917 498 720 Sabino Arana, 18 08028 BARCELONA Tlf. +34 934 098 480 Fax +34 934 098 490 http://www.comunycarse.com AVISO LEGAL La presente comunicación y sus anexos tiene como destinatario la persona a la que va dirigida, por lo que si usted lo recibe por error debe notificarlo al remitente y eliminarlo de su sistema, no pudiendo utilizarlo, total o parcialmente, para ningún fin. Su contenido puede tener información confidencial o protegida legalmente y únicamente expresa la opinión del remitente. El uso del correo electrónico vía internet no permite asegurar ni la confidencialidad de los mensajes ni su correcta recepción. En el caso de que el destinatario no consintiera la utilización del correo electrónico deberá ponerlo en nuestro conocimiento inmediatamente. DISCLAIMER This message and its attachments are intended exclusively for the named addressee. If you receive this message by mistake, please delete it immediately from your system and notify the sender. You may not use this message or any part of it for any purpose. The message may contain information that is confidential or protected by law, and any opinions expressed are those of the individual sender. Internet email guarantees neither the confidentiality nor the proper receipt of the message sent. If the addressee of this message does not consent to the use of internet e-mail, please inform us immediately. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at 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 at 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 OpenTelecom.IT cell: +39 347 266 56 18 -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.jpg Type: image/jpeg Size: 1630 bytes Desc: image001.jpg URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image002.jpg Type: image/jpeg Size: 1694 bytes Desc: image002.jpg URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image003.jpg Type: image/jpeg Size: 1628 bytes Desc: image003.jpg URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image004.jpg Type: image/jpeg Size: 1402 bytes Desc: image004.jpg URL: From admin at tedssupply.com Fri Jun 16 14:24:12 2017 From: admin at tedssupply.com (admin) Date: Fri, 16 Jun 2017 10:24:12 -0400 Subject: [Freeswitch-users] When calling out , freeswitch doesn't get DTMF ? In-Reply-To: References: <15c898b5d58.2779.b07ebdf329620b8089087c7205b03f01@xbipin.com> <310cf71c.8fc3.15c8ba9c4f6.Coremail.eastour@163.com> <2a18422f.ef8.15c99b7b9bb.Coremail.eastour@163.com> <6c11dd0f.c2c1.15c9c727dc7.Coremail.eastour@163.com> <2f48505d.d9c5.15ca1d28424.Coremail.eastour@163.com> <59401B92020000310000A34D@mail.tedssupply.com> <59412DB4020000310000A381@mail.tedssupply.com> <7d1e3f12.4750.15ca994d89d.Coremail.eastour@163.com> Message-ID: <5943B1CC020000310000A3F1@mail.tedssupply.com> I am finally onsite and have prepared the logs requested, but I am having a bit of trouble with Pastebin. Is there a max line count? I logged a good call, bad call, good sip trace, bad sip trace and including dialplan is about 17,000 lines -- too many? May just be a temp problem with server, will try again later. Thanks - James >>> Giovanni Maruzzelli 6/15/2017 05:04 AM >>> On 15 June 2017 at 04:28, chenyzhi wrote: the version is: FreeSWITCH Version 1.9.0+git~20170518T231917Z~a1fc18aee5~32bit (git a1fc18a 2017-05-18 23:19:17Z 32bit) the complete, since beginning to end, unedited, debug output of console when making a outgoing call which does not get DTMFs and the whole conf folder is in the attatchment. from the log, seems it does not read all the IVR messages correctly to you, it exits straight away... are you able to correctly hear all the IVR messages? also, can you take a SIP trace? (from console: "sofia global siptrace on") I suspect you have a NAT problem of some sort Also, I see you are on MASTER git, on Windows, and on 32 bit... Not sure this is supported... Have you has this problems with stable branch (1.6.x)? thx! 在 2017-06-15 00:42:11，"Giovanni Maruzzelli" 写道： On 14 June 2017 at 18:36, admin wrote: I don't mean to hijack the OP concern with my problem, I just wanted to reinforce I have seen this problem. And the problem is that the call originated through FS appears not to send user entered DTMF to the receiving phone. Let's help chenyzhi and then I'll take a turn. I just tested right now from console with: bgapi originate user/1011 5000 and originate user/1011 5000 and it works Please pastebin your FreeSWITCH version (eg, type "version" in console), your dialplan, your SIP profile, and the complete, since beginning to end, unedited, debug output of console when receiving a call which does not get DTMFs Maybe helping you we'll help him too - James >>> Giovanni Maruzzelli 6/14/2017 03:53 AM >>> Never heard such problems Please pastebin your dialplan, your SIP profile, and the complete, since beginning to end, unedited, debug output of console when receiving a call which does not get DTMFs sent from mobile cell: +39 347 266 56 18 Giovanni Maruzzelli OpenTelecom.IT On Jun 13, 2017 23:07, "admin" wrote: I am encountering the same issue. I am using an ESL call to api originate, with little else except getting in and out of the port, and the caller DTMF is failing to reach the called number. The called number auto answer attendant does not respond to DTMF, and a call to a test phone confirms no DTMF. This is a new upgrade from 1.2 to 1.6 and I don't know if this was an issue in 1.2, but my users never complained before 1.6 upgrade. Ideas?... >>> chenyzhi 06/13/17 3:32 PM >>> It's a one leg call .there is no b-leg. please make a test on your freeswitch box . just type the command "originate user/1001 5000" on the freeswitch console to see if your freeswitch instance can detect dtmf input. At 2017-06-12 22:57:17, "Luis Daniel Lucio Quiroz" wrote: check if you have transcoding, and if you do, check that dftm type-codec on leg b are compatible. -- Luis Daniel Lucio Quiroz CISSP, CISM, CISA Linux, VoIP and much more fun www.okay.com.mx Need LCR? Check out LCR for FusionPBX with FreeSWITCH Need Billing? Check out Billing for FusionPBX with FreeSWITCH On Mon, Jun 12, 2017 at 9:16 AM, chenyzhi wrote: I don't think It's the DTMF type , because when I dial 5000 from x-lite (which has registered to freeswitch as 1001) ,I can hear the voice and if I press any dtmf on x-lite, freeswitch can recieve the dtmfs. This means that the DTMF type is correct ,otherwise freeswitch coudn't have received the dtmfs; when I enter the command "originate user/1001 5000" at the freeswitch console , my xlite will ring ,and I answered ,I can hear the voice ,I press some dtmf,but freeswitch can NOT receive any dtmf. really weird. At 2017-06-12 17:37:49, "Giovanni Maruzzelli" wrote: you need to check the DTMF type, you probably are using the wrong one (info-inband-rfc2833), and for some reason they are not negotiated On 12 June 2017 at 02:32, chenyzhi wrote: I have read the logs ,but I didn't find any difference. Please make a test to see if this happens in your box. At 2017-06-10 06:49:34, "David Villasmil" wrote: Have you looked at the log? Bump the logging up and see what shows up... what you're seeing is very weird David On Fri, Jun 9, 2017 at 11:18 AM chenyzhi wrote: Hi call 5000 from x-lite , you can hear the IVR voice and if you press dtmf keys , freeswitch can receive the dtmf keys . but , if you enter the command "originate user/1001 5000" , x-lite will ring ,answer it ,you can hear the IVR voice , press some keys ,the freeswitch can NOT receive any dtmf , why? _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at 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 at 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 at 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 at 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 OpenTelecom.IT cell: +39 347 266 56 18 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at 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 at 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 at 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 at 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 at 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 at 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 OpenTelecom.IT cell: +39 347 266 56 18 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at 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 at 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 OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: From tculjaga at gmail.com Fri Jun 16 16:20:40 2017 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Fri, 16 Jun 2017 18:20:40 +0200 Subject: [Freeswitch-users] Call recording In-Reply-To: References: <7c47071f86ce436186b4ab8bf58de46c@w12mad02.comunycarse.com> <5323e6cfb2844e59a19957fad4e8044c@w12mad02.comunycarse.com> Message-ID: <6B44BE94-4FF7-4D38-98A5-F54BAB4A510A@gmail.com> I would not use lua, dialplan works great ;) Sent from my iPhone > On 16 Jun 2017, at 09:53, Sven Uszpelkat wrote: > > Hi Giovanni, > > Thank you for your help. We changed our script to something like this: > > session:answer() > while(session:ready() == true) do > test = session:recordFile("/usr/local/freeswitch/recordings/test.wav", 18000, 0, 300) > os.execute("sync"); > session:setAutoHangup(false) > session:hangup() > end > > As result we note an improvement primarily in short recordings (<1:30 min). These are now practically all complete. In longer recordings there are still losses and it seems that they are increasing with the duration of the recording. I’m not sure how to interpret this but to me it looks like this: with the sync call we achieved to write the buffer content to the file, however in longer calls there is remaining audio which hasn’t even been read to the buffer. Is that correct? If so then the recording function doesn’t ensure to read the remaining audio after hangup. (With tcdump we checked that all audio packets arrived before the BYE message) Does this mean that this behavior is by design? > > Best regards, > Sven > > De: Giovanni Maruzzelli [mailto:gmaruzz at gmail.com] > Enviado el: jueves, 15 de junio de 2017 13:17 > Para: Sven Uszpelkat > CC: FreeSWITCH Users Help > Asunto: Re: [Freeswitch-users] Call recording > > > > On 15 June 2017 at 12:50, Sven Uszpelkat wrote: > > 3.) We set the silence threshold to 0 because the documentation is not very clear how to disable the silence detection. We don’t want the recording to stop in response to a period of silence. The point is to record everything. > > > The silence_threshold determines what is considered silence, eg below what level of acoustic energy we state the stream is containing silence. Then, we wait for "how_many_silence_seconds" or until hangup before stopping recording. > > So, maybe you are right, and setting it to 0 will consider silence only when there is absolute silence in the stream, so for all practical purposes, until hangup. I have no mean to check source code now. > > On another hand, I can think at other possible causes for the premature end of the recorded file: maybe you move the file before it has been flushed by FreeSWITCH or by operating system? Maybe the hangup in the script interrupts the recording in the script and close the file descriptor before is flushed? (I am shooting in the dark) > > You can try to leave out those two lines, and test again. > > Also, you can insert a line that sync (flush) the filesystem before exiting, just to be sure. > > I would insert it after the while(session:ready()) > > A system(sync), or something similar will probably do. > > Hope this helps, > -giovanni > > > > > > > Best regards, > Sven > > De: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] En nombre de Giovanni Maruzzelli > Enviado el: jueves, 15 de junio de 2017 11:19 > Para: FreeSWITCH Users Help > Asunto: Re: [Freeswitch-users] Call recording > > > > On 15 June 2017 at 09:49, Sven Uszpelkat wrote: > Hello, > > We are using FreeSWITCH as a third-party recording application, i.e. we are receiving SIP calls with the complete audio of conversations taking place on another switch and we are saving this audio to a file. To achieve this we are using a simple script similar to this: > > session:answer() > while(session:ready() == true) do > test = session:recordFile("/usr/local/freeswitch/recordings/test.wav", 18000, 0, 300) > session:setAutoHangup(false) > session:hangup() > end > > This script will be invoked by the following dialplan: > > > > > > > > Basically it seems to work quite well, but sometimes there are missing audio at the end of the recorded file. Usually it’s only a few seconds, but sometimes it seems to be more. (It’s like the recording sometimes goes behind the real call and when the hangup event is received the remaining audio is discarded.) > > What could be the reason for this behavior? Is there something wrong with the script or is there a better way to achieve our goal? > > One first question come to my mind: why do you use a script here? A simple extension can do exactly the same, if you just want to record the session... > > https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools:+record_session > > Anyway, if you want to use the script, why you first > > session:setAutoHangup(false) > > and after that you > > session:hangup() > > ? > > Also, you made the silence_threshold equal 0 (zero). > Have you has the same problems using a silence_threshold of, let's say, 30 (thirty), like in documentation? ( https://freeswitch.org/confluence/display/FREESWITCH/Lua+API+Reference#LuaAPIReference-session:recordFile ) > > Hope this helps, > > -giovanni > > > > Many thanks in advance. > > Best regards, > > Sven Uszpelkat > Departamento I+D > Comunycarse Network Consultants, S.L. > > > > > Joaquín Turina, 2 > 28224 Pozuelo de Alarcón MADRID > Tlf. +34 917 498 700 > Fax +34 917 498 720 > > Sabino Arana, 18 > 08028 BARCELONA > Tlf. +34 934 098 480 > Fax +34 934 098 490 > > http://www.comunycarse.com > > AVISO LEGAL > La presente comunicación y sus anexos tiene como destinatario la persona a la que va dirigida, por lo que si usted lo recibe por error debe notificarlo al remitente y eliminarlo de su sistema, no pudiendo utilizarlo, total o parcialmente, para ningún fin. Su contenido puede tener información confidencial o protegida legalmente y únicamente expresa la opinión del remitente. El uso del correo electrónico vía internet no permite asegurar ni la confidencialidad de los mensajes ni su correcta recepción. En el caso de que el destinatario no consintiera la utilización del correo electrónico deberá ponerlo en nuestro conocimiento inmediatamente. > > DISCLAIMER > This message and its attachments are intended exclusively for the named addressee. If you receive this message by mistake, please delete it immediately from your system and notify the sender. You may not use this message or any part of it for any purpose. The message may contain information that is confidential or protected by law, and any opinions expressed are those of the individual sender. Internet email guarantees neither the confidentiality nor the proper receipt of the message sent. If the addressee of this message does not consent to the use of internet e-mail, please inform us immediately. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at 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 > OpenTelecom.IT > cell: +39 347 266 56 18 > > > > -- > > Sincerely, > > Giovanni Maruzzelli > OpenTelecom.IT > cell: +39 347 266 56 18 > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From soapee01.fs at stubbornroses.com Fri Jun 16 17:06:06 2017 From: soapee01.fs at stubbornroses.com (soapee01.fs at stubbornroses.com) Date: Fri, 16 Jun 2017 12:06:06 -0500 Subject: [Freeswitch-users] Missed Calls with Hunt Group In-Reply-To: <594304D6.7070207@stubbornroses.com> References: <594304D6.7070207@stubbornroses.com> Message-ID: <59440FFE.1040905@stubbornroses.com> All: Gill tested this for me on 1.6.17, and the hunt group does not show missed calls on phones that did not answer the call on 1.6.17. Regards, James On 6/15/2017 5:06 PM, soapee01.fs at stubbornroses.com wrote: > Hi, > > Here's the command that I'm running. > > originate user/102 at domain &bridge(user/100 at domain,user/101 at domain) > > > On FS version 1.2.22, if the user 100 answers the call, user 101 will > not see a missed call notification. > > On FS version FS 1.6.5 if the user 100 answers the call, user 101 > will show a missed call. > > Is there something in the docs I've missed? I'd really like to set it > back to the old way, but it would be really cool if there's a variable > somewhere I've missed that lets you choose the behavior. > > Thanks! > > James > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From adrian.worutowicz at esifrance.net Fri Jun 16 13:17:05 2017 From: adrian.worutowicz at esifrance.net (Adrian Worutowicz) Date: Fri, 16 Jun 2017 15:17:05 +0200 Subject: [Freeswitch-users] Build Problem in VS2015 In-Reply-To: <80c901d2e5f4$7e26f930$7a74eb90$@tollfreegateway.com> References: <001001d2e5bb$f4519cc0$dcf4d640$@worutowicz@esifrance.net> <80c901d2e5f4$7e26f930$7a74eb90$@tollfreegateway.com> Message-ID: <005101d2e6a2$d9b0d1a0$8d1274e0$@worutowicz@esifrance.net> I followed your steps, but unfortunately I got the same result. Probably I’m missing something in my VS install. Thanks a lot anyway De : FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de krice at tollfreegateway.com Envoyé : jeudi 15 juin 2017 18:29 À : 'FreeSWITCH Users Help' Objet : Re: [Freeswitch-users] Build Problem in VS2015 Not sure whats you are doing incorrect here, but I have just built master, I use the built in git bits with VS2015, and then drop to a command prompt (via the team explorer tab, select branches, right click the repo and select open command prompt) Then git pull, git clean -fdx, git reset –hard origin/master , git pull >From here back to the solution explorer open the FreeSWITCH.2015 solution file and build as normal I think you have something skewed there old ssl vs new ssl bits From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Adrian Worutowicz Sent: Thursday, June 15, 2017 4:44 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Build Problem in VS2015 Hello, I try to recompile FS in VS2015 without success. I took FS sources from git master. git config --global core.autocrlf false git clone https://stash.freeswitch.org/scm/fs/freeswitch.git /c/ESI/Components/FreeSwitch/ I have wix311 for VS2015 installed. For example it searches in folder 'openssl-1.0.2k' while only a folder 'openssl' exists. I tried to recompile mod_PortAudio, and I got c:\ESI\components\freeswitch\src\mod\endpoints\mod_portaudio\pablio.h(55): fatal error C1083: Impossible d'ouvrir le fichier include : 'portaudio.h' : No such file or directory. Indeed 'portaudio.h' does not exist. Plenty of other errors in the attached file. What do I miss? Thanks in advance, Adrian. -------------- next part -------------- An HTML attachment was scrubbed... URL: From rmundkowsky at ets.org Fri Jun 16 14:21:24 2017 From: rmundkowsky at ets.org (Mundkowsky, Robert) Date: Fri, 16 Jun 2017 14:21:24 +0000 Subject: [Freeswitch-users] Verto failover In-Reply-To: References: Message-ID: I think Sergey’s answer is good for you. I was just curious if Sergey knew if his solution supports active call failover as well, which I doubt, but I am not sure. Robert Mundkowsky From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ketan Kothari Sent: Friday, June 16, 2017 3:03 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Verto failover Hello Robert, I'm looking for fail-over for next call and web-interface as well as if one FreeSWITCH failed then verto continue working from other FreeSWITCH server. If you have any suggestion or reference link then please pass it will helpful for me. On Thu, Jun 15, 2017 at 7:21 PM, Mundkowsky, Robert > wrote: Verto is just frontend client side code (javascript), you can put it anywhere and point to a FreeSWITCH server anywhere else. I am curious what type of failover is handled by Sergey’s suggestion. Would this support an active call to continue? Or are you talking about failover for next call? Robert Mundkowsky From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sergey Safarov Sent: Thursday, June 15, 2017 7:17 AM To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Verto failover Yes. Balancing can be done using: 1) at dns level; 2) using haproxy daemon; 3) using nginx as proxy. чт, 15 июн. 2017 г. в 11:41, Ketan Kothari >: Can we setup Verto communicator User-Interface and FreeSWITCH on separate server? If yes ---> We have 2 FreeSWITCH servers and 1 User-Interface server. So how to load balance or fail-over of Verto? _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ________________________________ This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited. Thank you for your compliance. ________________________________ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ________________________________ This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited. Thank you for your compliance. ________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL: From s.safarov at gmail.com Fri Jun 16 17:38:08 2017 From: s.safarov at gmail.com (Sergey Safarov) Date: Fri, 16 Jun 2017 17:38:08 +0000 Subject: [Freeswitch-users] Verto failover In-Reply-To: References: Message-ID: NGINX example https://www.nginx.com/blog/nginx-nodejs-websockets-socketio/ Also you may look iptables solution https://www.webair.com/community/simple-stateful-load-balancer-with-iptables-and-nat/ чт, 15 июн. 2017 г. в 17:18, Ketan Kothari : > Hello Sergey, > > Thanks for your response. > > > > *3) using nginx as proxy.* > Could you please tell me some reference to do it. > > > On Thu, Jun 15, 2017 at 4:46 PM, Sergey Safarov > wrote: > >> Yes. >> Balancing can be done using: >> 1) at dns level; >> 2) using haproxy daemon; >> 3) using nginx as proxy. >> >> чт, 15 июн. 2017 г. в 11:41, Ketan Kothari : >> >>> Can we setup Verto communicator *User-Interface* and *FreeSWITCH* on >>> separate server? >>> If yes >>> >>> *--->* We have 2 FreeSWITCH servers and 1 User-Interface server. So how >>> to load balance or fail-over of Verto? >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at 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 at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From s.safarov at gmail.com Fri Jun 16 18:45:27 2017 From: s.safarov at gmail.com (Sergey Safarov) Date: Fri, 16 Jun 2017 18:45:27 +0000 Subject: [Freeswitch-users] Verto failover In-Reply-To: References: Message-ID: пт, 16 июн. 2017 г. в 20:40, Mundkowsky, Robert : > I think Sergey’s answer is good for you. > > > > I was just curious if Sergey knew if his solution supports active call > failover as well, which I doubt, but I am not sure. > Failover must be supported on on server side and on client side. I tested Verto client and may say failover is works on client side. Also i tested nginx in many cases, failover is works too. I not tested Verto failover on FreeSwitch side. -------------- next part -------------- An HTML attachment was scrubbed... URL: From s.safarov at gmail.com Fri Jun 16 18:52:28 2017 From: s.safarov at gmail.com (Sergey Safarov) Date: Fri, 16 Jun 2017 18:52:28 +0000 Subject: [Freeswitch-users] WebRTC and Carrier NAT ( CgNAT) In-Reply-To: References: Message-ID: Many carrier use this NAT schema. At home i have double NAT. 1) my home WiFi router 2) carrier cone NAT All works as expected. Tested WebRTC (sipML5) and Verto. пт, 16 июн. 2017 г. в 12:01, Tihomir Culjaga : > hello, > > does anyone have experience with WebRTC via carrier NAT ? > > client_LAN(192.168.x.y) carrer_prvate_WAN(10.x.x.x) > carrier_public_WAN(public_ip) <> internet > > > in this scenario, WebRTC calls using STUN only will not work. What about > TURN ? > > did anyone try that ? > > Regards, > Tihomir. > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From gregor at infomedia.si Fri Jun 16 19:19:07 2017 From: gregor at infomedia.si (Gregor Nanger) Date: Fri, 16 Jun 2017 19:19:07 +0000 Subject: [Freeswitch-users] Build Problem in VS2015 In-Reply-To: <594417cf.462ded0a.12ca4.f4aeSMTPIN_ADDED_BROKEN@mx.google.com> References: <80c901d2e5f4$7e26f930$7a74eb90$@tollfreegateway.com> <594417cf.462ded0a.12ca4.f4aeSMTPIN_ADDED_BROKEN@mx.google.com> Message-ID: I do not have problems compiling with visual Studio. Except for cloning, I use same command as stated in wiki. Then open in visual Studio and Build solution. On Fri, Jun 16, 2017, 19:39 Adrian Worutowicz < adrian.worutowicz at esifrance.net> wrote: > I followed your steps, but unfortunately I got the same result. > > Probably I’m missing something in my VS install. > > > > Thanks a lot anyway… > > > > > > > > *De :* FreeSWITCH-users [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *De la part de* > krice at tollfreegateway.com > *Envoyé :* jeudi 15 juin 2017 18:29 > *À :* 'FreeSWITCH Users Help' > *Objet :* Re: [Freeswitch-users] Build Problem in VS2015 > > > > Not sure whats you are doing incorrect here, but I have just built master, > I use the built in git bits with VS2015, and then drop to a command prompt > (via the team explorer tab, select branches, right click the repo and > select open command prompt) > > > > Then git pull, git clean -fdx, git reset –hard origin/master , git pull > > > > From here back to the solution explorer open the FreeSWITCH.2015 solution > file and build as normal… > > > > I think you have something skewed there old ssl vs new ssl bits > > > > *From:* FreeSWITCH-users [ > mailto:freeswitch-users-bounces at lists.freeswitch.org > ] *On Behalf Of *Adrian > Worutowicz > *Sent:* Thursday, June 15, 2017 4:44 AM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* [Freeswitch-users] Build Problem in VS2015 > > > > Hello, > > > > I try to recompile FS in VS2015 without success. > > I took FS sources from git master. > > > > git config --global core.autocrlf false > > git clone https://stash.freeswitch.org/scm/fs/freeswitch.git > /c/ESI/Components/FreeSwitch/ > > > > I have wix311 for VS2015 installed. > > > > For example it searches in folder 'openssl-1.0.2k' while only a folder > 'openssl' exists. > > > > I tried to recompile mod_PortAudio, and I got > c:\ESI\components\freeswitch\src\mod\endpoints\mod_portaudio\pablio.h(55): > fatal error C1083: Impossible d'ouvrir le fichier include : 'portaudio.h' : > No such file or directory. > > > > Indeed 'portaudio.h' does not exist. > > Plenty of other errors in the attached file. > > > > What do I miss? > > > > Thanks in advance, > > Adrian. > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Gregor Nanger *CTO* t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485 • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia • www.infomedia.si -------------- next part -------------- An HTML attachment was scrubbed... URL: From tculjaga at gmail.com Fri Jun 16 19:53:53 2017 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Fri, 16 Jun 2017 21:53:53 +0200 Subject: [Freeswitch-users] WebRTC and Carrier NAT ( CgNAT) In-Reply-To: References: Message-ID: did you try restricted ( address, port ) cone NAT and Symmetric NAT as well ? i guess for Symmetric we do need a TURN server... not sure about other NAT types/Methods of translation On 16 June 2017 at 20:52, Sergey Safarov wrote: > Many carrier use this NAT schema. > At home i have double NAT. > 1) my home WiFi router > 2) carrier cone NAT > > All works as expected. Tested WebRTC (sipML5) and Verto. > > > > пт, 16 июн. 2017 г. в 12:01, Tihomir Culjaga : > >> hello, >> >> does anyone have experience with WebRTC via carrier NAT ? >> >> client_LAN(192.168.x.y) carrer_prvate_WAN(10.x.x.x) >> carrier_public_WAN(public_ip) <> internet >> >> >> in this scenario, WebRTC calls using STUN only will not work. What about >> TURN ? >> >> did anyone try that ? >> >> Regards, >> Tihomir. >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian at freeswitch.org Fri Jun 16 21:22:26 2017 From: brian at freeswitch.org (Brian West) Date: Fri, 16 Jun 2017 16:22:26 -0500 Subject: [Freeswitch-users] WebRTC and Carrier NAT ( CgNAT) In-Reply-To: References: Message-ID: If you're going to be doing Carrier NAT, You should use the Carrier NAT Network Range. That would be 100.64.0.0/10 :) On Fri, Jun 16, 2017 at 2:53 PM, Tihomir Culjaga wrote: > did you try restricted ( address, port ) cone NAT and Symmetric NAT as > well ? > > i guess for Symmetric we do need a TURN server... not sure about other NAT > types/Methods of translation > > On 16 June 2017 at 20:52, Sergey Safarov wrote: > >> Many carrier use this NAT schema. >> At home i have double NAT. >> 1) my home WiFi router >> 2) carrier cone NAT >> >> All works as expected. Tested WebRTC (sipML5) and Verto. >> >> >> >> пт, 16 июн. 2017 г. в 12:01, Tihomir Culjaga : >> >>> hello, >>> >>> does anyone have experience with WebRTC via carrier NAT ? >>> >>> client_LAN(192.168.x.y) carrer_prvate_WAN(10.x.x.x) >>> carrier_public_WAN(public_ip) <> internet >>> >>> >>> in this scenario, WebRTC calls using STUN only will not work. What about >>> TURN ? >>> >>> did anyone try that ? >>> >>> Regards, >>> Tihomir. >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at 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 at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Book a phone call (CST) Allison prompts for FreeSWITCH: *https://www.gofundme.com/allison-prompts-for-freeswitch* Got Bugs? Report them here ! | Reddit: /r/freeswitch *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: From gmaruzz at gmail.com Sat Jun 17 08:30:38 2017 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Sat, 17 Jun 2017 10:30:38 +0200 Subject: [Freeswitch-users] Call recording In-Reply-To: <6B44BE94-4FF7-4D38-98A5-F54BAB4A510A@gmail.com> References: <7c47071f86ce436186b4ab8bf58de46c@w12mad02.comunycarse.com> <5323e6cfb2844e59a19957fad4e8044c@w12mad02.comunycarse.com> <6B44BE94-4FF7-4D38-98A5-F54BAB4A510A@gmail.com> Message-ID: On 16 June 2017 at 18:20, Tihomir Culjaga wrote: > I would not use lua, dialplan works great ;) > > Sent from my iPhone > > On 16 Jun 2017, at 09:53, Sven Uszpelkat wrote: > > Hi Giovanni, > > > > Thank you for your help. We changed our script to something like this: > > > > session:answer() > > while(session:ready() == true) do > > test = session:recordFile("/usr/local/freeswitch/recordings/test.wav", > 18000, 0, 300) > > os.execute("sync"); > > session:setAutoHangup(false) > > session:hangup() > > end > > I would put an "if-end" instead of a "while-do", and put sync outside the "ifend", setautohangup before the ifend, hangup after the ifend you can also use sync-sleeponesecond-syncagain (after the ifend block), just to be very proactive :) -giovanni > > > As result we note an improvement primarily in short recordings (<1:30 > min). These are now practically all complete. In longer recordings there > are still losses and it seems that they are increasing with the duration of > the recording. I’m not sure how to interpret this but to me it looks like > this: with the sync call we achieved to write the buffer content to the > file, however in longer calls there is remaining audio which hasn’t even > been read to the buffer. Is that correct? If so then the recording function > doesn’t ensure to read the remaining audio after hangup. (With tcdump we > checked that all audio packets arrived before the BYE message) Does this > mean that this behavior is by design? > > > > Best regards, > > Sven > > > > *De:* Giovanni Maruzzelli [mailto:gmaruzz at gmail.com ] > *Enviado el:* jueves, 15 de junio de 2017 13:17 > *Para:* Sven Uszpelkat > *CC:* FreeSWITCH Users Help > *Asunto:* Re: [Freeswitch-users] Call recording > > > > > > > > On 15 June 2017 at 12:50, Sven Uszpelkat wrote: > > > > 3.) We set the silence threshold to 0 because the documentation is not > very clear how to disable the silence detection. We don’t want the > recording to stop in response to a period of silence. The point is to > record everything. > > > > The silence_threshold determines what is considered silence, eg below what > level of acoustic energy we state the stream is containing silence. Then, > we wait for "how_many_silence_seconds" or until hangup before stopping > recording. > > So, maybe you are right, and setting it to 0 will consider silence only > when there is absolute silence in the stream, so for all practical > purposes, until hangup. I have no mean to check source code now. > > On another hand, I can think at other possible causes for the premature > end of the recorded file: maybe you move the file before it has been > flushed by FreeSWITCH or by operating system? Maybe the hangup in the > script interrupts the recording in the script and close the file descriptor > before is flushed? (I am shooting in the dark) > > You can try to leave out those two lines, and test again. > > Also, you can insert a line that sync (flush) the filesystem before > exiting, just to be sure. > > I would insert it after the while(session:ready()) > > A system(sync), or something similar will probably do. > > > > Hope this helps, > > -giovanni > > > > > > > > > > Best regards, > > Sven > > > > *De:* FreeSWITCH-users [mailto:freeswitch-users- > bounces at lists.freeswitch.org > ] *En nombre de *Giovanni > Maruzzelli > *Enviado el:* jueves, 15 de junio de 2017 11:19 > *Para:* FreeSWITCH Users Help > *Asunto:* Re: [Freeswitch-users] Call recording > > > > > > > > On 15 June 2017 at 09:49, Sven Uszpelkat wrote: > > Hello, > > > > We are using FreeSWITCH as a third-party recording application, i.e. we > are receiving SIP calls with the complete audio of conversations taking > place on another switch and we are saving this audio to a file. To achieve > this we are using a simple script similar to this: > > > > session:answer() > > while(session:ready() == true) do > > test = session:recordFile("/usr/local/freeswitch/recordings/test.wav", > 18000, 0, 300) > > session:setAutoHangup(false) > > session:hangup() > > end > > > > This script will be invoked by the following dialplan: > > > > > > > > > > > > > > > > Basically it seems to work quite well, but sometimes there are missing > audio at the end of the recorded file. Usually it’s only a few seconds, > but sometimes it seems to be more. (It’s like the recording sometimes goes > behind the real call and when the hangup event is received the remaining > audio is discarded.) > > > > What could be the reason for this behavior? Is there something wrong with > the script or is there a better way to achieve our goal? > > > > One first question come to my mind: why do you use a script here? A simple > extension can do exactly the same, if you just want to record the session... > > https://freeswitch.org/confluence/display/FREESWITCH/ > mod_dptools:+record_session > > Anyway, if you want to use the script, why you first > > session:setAutoHangup(false) > > and after that you > > > > session:hangup() > > > > ? > > > > Also, you made the silence_threshold equal 0 (zero). > Have you has the same problems using a silence_threshold of, let's say, 30 > (thirty), like in documentation? ( https://freeswitch.org/ > confluence/display/FREESWITCH/Lua+API+Reference#LuaAPIReference-session: > recordFile ) > > > > Hope this helps, > > -giovanni > > > > > > > Many thanks in advance. > > > > Best regards, > > > > Sven Uszpelkat > Departamento I+D > Comunycarse Network Consultants, S.L. > > > > > > > > > > > > Joaquín Turina, 2 > 28224 Pozuelo de Alarcón MADRID > Tlf. +34 917 498 700 <+34%20917%2049%2087%2000> > Fax +34 917 498 720 <+34%20917%2049%2087%2020> > > Sabino Arana, 18 > 08028 BARCELONA > Tlf. +34 934 098 480 <+34%20934%2009%2084%2080> > Fax +34 934 098 490 <+34%20934%2009%2084%2090> > > http://www.comunycarse.com > > > AVISO LEGAL > La presente comunicación y sus anexos tiene como destinatario la persona a > la que va dirigida, por lo que si usted lo recibe por error debe > notificarlo al remitente y eliminarlo de su sistema, no pudiendo > utilizarlo, total o parcialmente, para ningún fin. Su contenido puede tener > información confidencial o protegida legalmente y únicamente expresa la > opinión del remitente. El uso del correo electrónico vía internet no > permite asegurar ni la confidencialidad de los mensajes ni su correcta > recepción. En el caso de que el destinatario no consintiera la utilización > del correo electrónico deberá ponerlo en nuestro conocimiento > inmediatamente. > > DISCLAIMER > This message and its attachments are intended exclusively for the named > addressee. If you receive this message by mistake, please delete it > immediately from your system and notify the sender. You may not use this > message or any part of it for any purpose. The message may contain > information that is confidential or protected by law, and any opinions > expressed are those of the individual sender. Internet email guarantees > neither the confidentiality nor the proper receipt of the message sent. If > the addressee of this message does not consent to the use of internet > e-mail, please inform us immediately. > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at 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 > OpenTelecom.IT > cell: +39 347 266 56 18 > > > > > -- > > > Sincerely, > > Giovanni Maruzzelli > OpenTelecom.IT > cell: +39 347 266 56 18 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at 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 OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: From gmaruzz at gmail.com Sat Jun 17 08:32:17 2017 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Sat, 17 Jun 2017 10:32:17 +0200 Subject: [Freeswitch-users] When calling out , freeswitch doesn't get DTMF ? In-Reply-To: <7611bb49.72be.15caf10d2a0.Coremail.eastour@163.com> References: <15c898b5d58.2779.b07ebdf329620b8089087c7205b03f01@xbipin.com> <310cf71c.8fc3.15c8ba9c4f6.Coremail.eastour@163.com> <2a18422f.ef8.15c99b7b9bb.Coremail.eastour@163.com> <6c11dd0f.c2c1.15c9c727dc7.Coremail.eastour@163.com> <2f48505d.d9c5.15ca1d28424.Coremail.eastour@163.com> <59401B92020000310000A34D@mail.tedssupply.com> <59412DB4020000310000A381@mail.tedssupply.com> <7d1e3f12.4750.15ca994d89d.Coremail.eastour@163.com> <7611bb49.72be.15caf10d2a0.Coremail.eastour@163.com> Message-ID: On 16 June 2017 at 06:02, chenyzhi wrote: > Yes ,I can hear all the IVR prompt voices correctly. > > I don't think it's a NAT problem ,because both the x-lite and the > freeswitch are in the same LAN. > > The sip trace log is in the attatchment. Thank you. > > PS I tested this on another freeswitch box ,version: > FreeSWITCH Version 1.6.16+git~20170403T142423Z~e6d643b29c~32bit (git > e6d643b 2017-04-03 14:24:23Z 32bit) > It can detect dtmf on outgoing calls. Maybe this only happens on > FreeSWITCH Version 1.9.0+git~20170518T231917Z~a1fc18aee5~32bit (git > a1fc18a 2017-05-18 23:19:17Z 32bit) > then use the stable version, and open a jira for this issue citing the master version you are using -------------- next part -------------- An HTML attachment was scrubbed... URL: From richard.mace at gmail.com Sat Jun 17 09:47:18 2017 From: richard.mace at gmail.com (Richard Mace) Date: Sat, 17 Jun 2017 10:47:18 +0100 Subject: [Freeswitch-users] Showing caller number In-Reply-To: References: Message-ID: Hi, I have a situation where I have users dialling a number on my freeswitch system starting with 01794 that then immediately calls a mobile number. When the mobile rings, it looks like the 01794 number is calling the number. How can I adjust the dial string so that when I call the mobile from freeswitch, it looks like like original number calling the 01794 number is ringing the mobile directly? Thanks Richard -------------- next part -------------- An HTML attachment was scrubbed... URL: From gmaruzz at gmail.com Sat Jun 17 11:01:16 2017 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Sat, 17 Jun 2017 13:01:16 +0200 Subject: [Freeswitch-users] Showing caller number In-Reply-To: References: Message-ID: https://freeswitch.org/confluence/display/FREESWITCH/Caller+ID+Privacy#CallerIDPrivacy-SettingCIDMethod On 17 June 2017 at 11:47, Richard Mace wrote: > Hi, > > I have a situation where I have users dialling a number on my freeswitch > system starting with 01794 that then immediately calls a mobile number. > When the mobile rings, it looks like the 01794 number is calling the > number. > > How can I adjust the dial string so that when I call the mobile from > freeswitch, it looks like like original number calling the 01794 number is > ringing the mobile directly? > > Thanks > > Richard > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at 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 OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: From lconroy at insensate.co.uk Sat Jun 17 11:26:28 2017 From: lconroy at insensate.co.uk (Lawrence Conroy) Date: Sat, 17 Jun 2017 12:26:28 +0100 Subject: [Freeswitch-users] Showing caller number In-Reply-To: References: Message-ID: <6061D25F-98A7-480E-8DCB-36B680213847@insensate.co.uk> Hi there, so, basically, you want to fake the outgoing CLI. ISTM that this is NOT a freeSwitch problem. freeswitch -could- send a different caller ID when outcalling, but you're using some supplier to call out to the mobile via the PSTN. Typically, that supplier will either trust that you "own" the calling number you're presenting (customer provided network screened), or will not want OfCom jumping up & down on their head and so put what they believe is your number in the SS#7 message they send to the mobile operator in the outcall (network provided calling number). Note that the supplier is supposed to at least screen whatever number you provide. It's their responsibility. Given that OfCom tends to be unhappy with people faking calling numbers, this is a hard problem. => I suspect that it is not something that freeSwitch can fix with UK based outcalls. best regards, Lawrence Conroy (who used to have an 01794 833xxx number) On 17 Jun 2017, at 10:47, Richard Mace wrote: > Hi, > > I have a situation where I have users dialling a number on my freeswitch system starting with 01794 that then immediately calls a mobile number. > When the mobile rings, it looks like the 01794 number is calling the number. > > How can I adjust the dial string so that when I call the mobile from freeswitch, it looks like like original number calling the 01794 number is ringing the mobile directly? > > Thanks > > Richard From infos at madovsky.org Sat Jun 17 11:55:07 2017 From: infos at madovsky.org (Madovsky) Date: Sat, 17 Jun 2017 04:55:07 -0700 Subject: [Freeswitch-users] timer not properly configured Message-ID: <70e56fd8-4c40-53e1-2de2-bd8b879daad2@madovsky.org> Hi all, last today git gives switch_core_timer.c:117 Timer is not properly configured everytime a call is hangup. show timer gives type,name,ikey timer,soft,CORE_SOFTTIMER_MODULE 1 total. Thanks F From stefan at fuhrmann.homedns.org Sat Jun 17 14:55:33 2017 From: stefan at fuhrmann.homedns.org (Stefan Fuhrmann) Date: Sat, 17 Jun 2017 16:55:33 +0200 Subject: [Freeswitch-users] enable Portal, error 404 Message-ID: <8779966.9K6fzknjRZ@stefan-ubu> Hello all, Im new to freeswitch and have to ask, how can I enable the portal? I installed the debian installation and followed the instruction from wiki to enable: https://wiki.freeswitch.org/wiki/Freeswitch_Portal It is based on mod_xml_rpc, the module is built by default but not loaded, so you just need to load it (un-comment it in conf/autoload_configs/ modules.conf.xml) load mod_xml_rpc When I trying to access ip:8080/portal/index.html after login Im getting: error 404 What Im missing? Can somone help? Tia Stefan From ksh.sip at gmail.com Sat Jun 17 16:16:09 2017 From: ksh.sip at gmail.com (Gauri Kshirsagar) Date: Sat, 17 Jun 2017 21:46:09 +0530 Subject: [Freeswitch-users] Adding video to audio call Message-ID: Hi, I am using Freeswitch version 1.9.0. I can make video calls. But when I try to add video to audio call it does not work. A makes audio call to B. Call is established. A adds video . ReINVITE sent to freeswitch has video added in SDP.Freeswitch sends 200 OK response for this INVITE to A which has video in SDP. But there is no ReINVITE being sent to B. I tried enabling renegotiate-codec-on-reinvite in vars.xml and also internal.xml Is this supported? If so is some other configuration required. Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: From anthony.minessale at gmail.com Sat Jun 17 17:29:31 2017 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 17 Jun 2017 17:29:31 +0000 Subject: [Freeswitch-users] timer not properly configured In-Reply-To: <70e56fd8-4c40-53e1-2de2-bd8b879daad2@madovsky.org> References: <70e56fd8-4c40-53e1-2de2-bd8b879daad2@madovsky.org> Message-ID: Jira jira jira On Sat, Jun 17, 2017 at 7:55 AM Madovsky wrote: > Hi all, > > last today git gives > > switch_core_timer.c:117 Timer is not properly configured > > everytime a call is hangup. > > show timer gives > > type,name,ikey > timer,soft,CORE_SOFTTIMER_MODULE > > 1 total. > > > Thanks > > F > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at 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 ♬ @anthmfs ♬ @FreeSWITCH ♬ ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ * ClueCon Weekly Development Call ☎ sip:888 at conference.freeswitch.org ☎ +19193869900 https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA -------------- next part -------------- An HTML attachment was scrubbed... URL: From s.safarov at gmail.com Sat Jun 17 18:39:25 2017 From: s.safarov at gmail.com (Sergey Safarov) Date: Sat, 17 Jun 2017 18:39:25 +0000 Subject: [Freeswitch-users] Adding video to audio call In-Reply-To: References: Message-ID: Requred to enable proxy media. сб, 17 июня 2017 г., 19:20 Gauri Kshirsagar : > Hi, > > I am using Freeswitch version 1.9.0. I can make video calls. But when I > try to add video to audio call it does not work. > > A makes audio call to B. Call is established. A adds video . > > ReINVITE sent to freeswitch has video added in SDP.Freeswitch sends 200 > OK response for this INVITE to A which has video in SDP. But there is no > ReINVITE being sent to B. > > I tried enabling renegotiate-codec-on-reinvite in vars.xml and also > internal.xml > > Is this supported? If so is some other configuration required. > > Regards, > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From mateo.felipe05 at gmail.com Sat Jun 17 15:34:39 2017 From: mateo.felipe05 at gmail.com (Felipe Mateo) Date: Sat, 17 Jun 2017 11:34:39 -0400 Subject: [Freeswitch-users] Fwd: No video playback with mod_av In-Reply-To: References: Message-ID: Hi all, I am trying to use mod_av to playback application with video support; but it does not show video output. It only shows video when dialing sample conference (canvas and screen). I also searched confluence but there is no documentation for mod_av Thanks in advance -------------- next part -------------- An HTML attachment was scrubbed... URL: From bipin at xbipin.com Sun Jun 18 06:02:10 2017 From: bipin at xbipin.com (Bipin Patel) Date: Sun, 18 Jun 2017 10:02:10 +0400 Subject: [Freeswitch-users] Adding video to audio call In-Reply-To: References: Message-ID: hi, also i believe there is a bug request open on jira relating to this Regards, Bipin ------------------------------------------------------------------------ -------- Original Message -------- Subject: Re: [Freeswitch-users] Adding video to audio call From: Sergey Safarov To: FreeSWITCH Users Help Date: 6/17/2017, 10:39:25 PM > > Requred to enable proxy media. > > > сб, 17 июня 2017 г., 19:20 Gauri Kshirsagar >: > > Hi, > > I am using Freeswitch version 1.9.0. I can make video calls. But > when I try to add video to audio call it does not work. > > A makes audio call to B. Call is established. A adds video . > > ReINVITE sent to freeswitch has video added in SDP.Freeswitch > sends 200 OK response for this INVITE to A which has video in SDP. > But there is no ReINVITE being sent to B. > > I tried enabling renegotiate-codec-on-reinvite in vars.xml and > also internal.xml > > Is this supported? If so is some other configuration required. > > Regards, > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From lexxua at gmail.com Sun Jun 18 18:59:06 2017 From: lexxua at gmail.com (Volodymyr Fedorov) Date: Sun, 18 Jun 2017 20:59:06 +0200 Subject: [Freeswitch-users] Debian Stretch In-Reply-To: <733A7BE7-6E66-4BD8-8736-B6DDBFE695A4@jerris.com> References: <733A7BE7-6E66-4BD8-8736-B6DDBFE695A4@jerris.com> Message-ID: Hi Michael, so from today Stretch is current stable it will be really cool to have packages from freeswitch repository . Thanks! On Tue, Jun 13, 2017 at 4:14 AM, Michael Jerris wrote: > announcements will come out when we have real dates. > > On Jun 12, 2017, at 9:32 PM, Peter Rex wrote: > > Thanks Michael. Hate to do this to you, but is there an estimate on 1.8 > timeframe? Mailing list shows people were talking about configs and feature > requests in January, but can't see much else. Maybe I'm not looking in the > right place. > > On Mon, Jun 12, 2017 at 6:44 PM, Michael Jerris wrote: > >> Stretch won’t build yet. I’ll have some patches over the next few weeks >> to fix that. 1.8 when released will likely target Stretch as its primary >> but still a bunch of testing to do. The patches to fix build for stretch >> will go back into 1.6 branch, once they are complete and tested. >> >> >> On Jun 12, 2017, at 8:28 PM, Peter Rex wrote: >> >> Stretch is the new stable on Saturday. I've looked through Confluence and >> the mailing lists but I can't find anything relevant. I see interesting >> possibilities at http://files.freeswitch.org/repo/deb, but I thought I >> would ask the mailing list if there's a plan yet to add or move the >> _production_ build to Stretch. >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Best regards, Volodymyr -------------- next part -------------- An HTML attachment was scrubbed... URL: From joelists at tm.net.uk Sun Jun 18 22:52:08 2017 From: joelists at tm.net.uk (Joseph Waite) Date: Sun, 18 Jun 2017 23:52:08 +0100 Subject: [Freeswitch-users] Call Dropping Message-ID: <222F1F16-373C-484A-B290-200CB13F2CAD@tm.net.uk> Hi Guys Using FreeSwitch with Radius linked to JeraSoft VCS billing system. I am sending a Call from a SIPP originator, through the FreeSwitch box and back out to another SIPP terminator scenario. The call goes through ok, everything happens as it should, however the call immediately drops, I have done egrep’s of both sides of the call and the BYE is defiantly coming from Freeswitch for some reason but I cannot work out why. Anyone any ideas? I am attaching the FreeSwitch logs plus the egrep’s If I register zipper on my laptop to FS and make a call works fine. 2017-06-18 22:26:34.199519 [NOTICE] switch_channel.c:1104 New Channel sofia/external/sipp at 185.35.228.51 :5060 [15021010-8f64-439f-8dbb-1afe090c44a5] 2017-06-18 22:26:34.199519 [DEBUG] switch_core_state_machine.c:584 (sofia/external/sipp at 185.35.228.51 :5060) Running State Change CS_NEW (Cur 1 Tot 67) 2017-06-18 22:26:34.199519 [DEBUG] sofia.c:9837 sofia/external/sipp at 185.35.228.51 :5060 receiving invite from 185.35.228.51:5060 version: 1.6.18 git 6e79667 2017-06-12 21:14:49Z 64bit 2017-06-18 22:26:34.219492 [DEBUG] sofia.c:7048 Channel sofia/external/sipp at 185.35.228.51 :5060 entering state [received][100] 2017-06-18 22:26:34.219492 [DEBUG] sofia.c:7058 Remote SDP: v=0 o=user1 53655765 2353687637 IN IP4 185.35.228.51 s=- c=IN IP4 185.35.228.51 t=0 0 m=audio 6000 RTP/AVP 0 a=rtpmap:0 PCMU/8000 2017-06-18 22:26:34.219492 [DEBUG] sofia.c:7450 (sofia/external/sipp at 185.35.228.51 :5060) State Change CS_NEW -> CS_INIT 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:603 (sofia/external/sipp at 185.35.228.51 :5060) State NEW 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:584 (sofia/external/sipp at 185.35.228.51 :5060) Running State Change CS_INIT (Cur 1 Tot 67) 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:627 (sofia/external/sipp at 185.35.228.51 :5060) State INIT 2017-06-18 22:26:34.219492 [DEBUG] mod_sofia.c:90 sofia/external/sipp at 185.35.228.51 :5060 SOFIA INIT 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:40 sofia/external/sipp at 185.35.228.51 :5060 Standard INIT 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:48 (sofia/external/sipp at 185.35.228.51 :5060) State Change CS_INIT -> CS_ROUTING 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:627 (sofia/external/sipp at 185.35.228.51 :5060) State INIT going to sleep 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:584 (sofia/external/sipp at 185.35.228.51 :5060) Running State Change CS_ROUTING (Cur 1 Tot 67) 2017-06-18 22:26:34.219492 [DEBUG] switch_channel.c:2249 (sofia/external/sipp at 185.35.228.51 :5060) Callstate Change DOWN -> RINGING 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:643 (sofia/external/sipp at 185.35.228.51 :5060) State ROUTING 2017-06-18 22:26:34.219492 [DEBUG] mod_sofia.c:143 sofia/external/sipp at 185.35.228.51 :5060 SOFIA ROUTING 2017-06-18 22:26:34.219492 [ERR] mod_xml_radius.c:933 Result of true match: 185.35.228.40 == ^185\.35\.229\.30 2017-06-18 22:26:34.219492 [INFO] mod_xml_radius.c:986 mod_xml_radius: Accounting Start success 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:236 sofia/external/sipp at 185.35.228.51 :5060 Standard ROUTING 2017-06-18 22:26:34.219492 [INFO] mod_dialplan_xml.c:637 Processing sipp ->441554555666 in context public Dialplan: sofia/external/sipp at 185.35.228.51 :5060 parsing [public->unloop] continue=false Dialplan: sofia/external/sipp at 185.35.228.51 :5060 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/external/sipp at 185.35.228.51 :5060 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/external/sipp at 185.35.228.51 :5060 parsing [public->outside_call] continue=true Dialplan: sofia/external/sipp at 185.35.228.51 :5060 Absolute Condition [outside_call] Dialplan: sofia/external/sipp at 185.35.228.51 :5060 Action set(outside_call=true) Dialplan: sofia/external/sipp at 185.35.228.51 :5060 Action export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) Dialplan: sofia/external/sipp at 185.35.228.51 :5060 parsing [public->call_debug] continue=true Dialplan: sofia/external/sipp at 185.35.228.51 :5060 Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never Dialplan: sofia/external/sipp at 185.35.228.51 :5060 parsing [public->rejections] continue=false Dialplan: sofia/external/sipp at 185.35.228.51 :5060 Regex (FAIL) [rejections] ${radius_auth_result}() =~ /2/ break=on-false Dialplan: sofia/external/sipp at 185.35.228.51 :5060 parsing [public->timedouts] continue=false Dialplan: sofia/external/sipp at 185.35.228.51 :5060 Regex (FAIL) [timedouts] ${radius_auth_result}() =~ /1/ break=on-false Dialplan: sofia/external/sipp at 185.35.228.51 :5060 parsing [public->JeraSoft VCS Routing] continue=false Dialplan: sofia/external/sipp at 185.35.228.51 :5060 Regex (PASS) [JeraSoft VCS Routing] destination_number(441554555666) =~ /^(.+)$/ break=on-false Dialplan: sofia/external/sipp at 185.35.228.51 :5060 Action export(nolocal:h323-call-origin=originate) Dialplan: sofia/external/sipp at 185.35.228.51 :5060 Action set(sip_h_X-accountcode=${accountcode}) Dialplan: sofia/external/sipp at 185.35.228.51 :5060 Action set(call_direction=outbound) Dialplan: sofia/external/sipp at 185.35.228.51 :5060 Action set(hangup_after_bridge=true) Dialplan: sofia/external/sipp at 185.35.228.51 :5060 Action set(continue_on_fail=true) Dialplan: sofia/external/sipp at 185.35.228.51 :5060 Action set(inherit_codec=true) Dialplan: sofia/external/sipp at 185.35.228.51 :5060 Action set(call_timeout=20) Dialplan: sofia/external/sipp at 185.35.228.51 :5060 Action set(fail_on_single_reject=USER_BUSY) Dialplan: sofia/external/sipp at 185.35.228.51 :5060 Action set(origination_caller_id_name=${sip_req_user}) Dialplan: sofia/external/sipp at 185.35.228.51 :5060 Action set(origination_caller_id_number=${sip_from_user}) Dialplan: sofia/external/sipp at 185.35.228.51 :5060 Action set(execute_on_answer=sched_hangup +${h323-credit-time} alloted_timeout) Dialplan: sofia/external/sipp at 185.35.228.51 :5060 Action bridge({sip_invite_from_uri=sip:${sip_from_user}@${sip_network_ip}}sofia/external/${destination_number}@185.35.229.30:5060) Dialplan: sofia/external/sipp at 185.35.228.51 :5060 Action hangup(${bridge_hangup_cause}) 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:286 (sofia/external/sipp at 185.35.228.51 :5060) State Change CS_ROUTING -> CS_EXECUTE 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:643 (sofia/external/sipp at 185.35.228.51 :5060) State ROUTING going to sleep 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:584 (sofia/external/sipp at 185.35.228.51 :5060) Running State Change CS_EXECUTE (Cur 1 Tot 67) 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:650 (sofia/external/sipp at 185.35.228.51 :5060) State EXECUTE 2017-06-18 22:26:34.219492 [DEBUG] mod_sofia.c:198 sofia/external/sipp at 185.35.228.51 :5060 SOFIA EXECUTE 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:328 sofia/external/sipp at 185.35.228.51 :5060 Standard EXECUTE EXECUTE sofia/external/sipp at 185.35.228.51 :5060 set(outside_call=true) 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET sofia/external/sipp at 185.35.228.51 :5060 [outside_call]=[true] EXECUTE sofia/external/sipp at 185.35.228.51 :5060 export(RFC2822_DATE=Sun, 18 Jun 2017 22:26:34 +0100) 2017-06-18 22:26:34.219492 [DEBUG] switch_channel.c:1296 EXPORT (export_vars) [RFC2822_DATE]=[Sun, 18 Jun 2017 22:26:34 +0100] EXECUTE sofia/external/sipp at 185.35.228.51 :5060 export(nolocal:h323-call-origin=originate) 2017-06-18 22:26:34.219492 [DEBUG] switch_channel.c:1296 EXPORT (export_vars) (REMOTE ONLY) [h323-call-origin]=[originate] EXECUTE sofia/external/sipp at 185.35.228.51 :5060 set(sip_h_X-accountcode=) 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET sofia/external/sipp at 185.35.228.51 :5060 [sip_h_X-accountcode]=[UNDEF] EXECUTE sofia/external/sipp at 185.35.228.51 :5060 set(call_direction=outbound) 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET sofia/external/sipp at 185.35.228.51 :5060 [call_direction]=[outbound] EXECUTE sofia/external/sipp at 185.35.228.51 :5060 set(hangup_after_bridge=true) 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET sofia/external/sipp at 185.35.228.51 :5060 [hangup_after_bridge]=[true] EXECUTE sofia/external/sipp at 185.35.228.51 :5060 set(continue_on_fail=true) 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET sofia/external/sipp at 185.35.228.51 :5060 [continue_on_fail]=[true] EXECUTE sofia/external/sipp at 185.35.228.51 :5060 set(inherit_codec=true) 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET sofia/external/sipp at 185.35.228.51 :5060 [inherit_codec]=[true] EXECUTE sofia/external/sipp at 185.35.228.51 :5060 set(call_timeout=20) 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET sofia/external/sipp at 185.35.228.51 :5060 [call_timeout]=[20] EXECUTE sofia/external/sipp at 185.35.228.51 :5060 set(fail_on_single_reject=USER_BUSY) 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET sofia/external/sipp at 185.35.228.51 :5060 [fail_on_single_reject]=[USER_BUSY] EXECUTE sofia/external/sipp at 185.35.228.51 :5060 set(origination_caller_id_name=441554555666) 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET sofia/external/sipp at 185.35.228.51 :5060 [origination_caller_id_name]=[441554555666] EXECUTE sofia/external/sipp at 185.35.228.51 :5060 set(origination_caller_id_number=sipp) 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET sofia/external/sipp at 185.35.228.51 :5060 [origination_caller_id_number]=[sipp] EXECUTE sofia/external/sipp at 185.35.228.51 :5060 set(execute_on_answer=sched_hangup + alloted_timeout) 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET sofia/external/sipp at 185.35.228.51 :5060 [execute_on_answer]=[sched_hangup + alloted_timeout] EXECUTE sofia/external/sipp at 185.35.228.51 :5060 bridge({sip_invite_from_uri=sip:sipp at 185.35.228.51}sofia/external/441554555666 at 185.35.229.30 :5060) 2017-06-18 22:26:34.219492 [DEBUG] switch_channel.c:1250 sofia/external/sipp at 185.35.228.51 :5060 EXPORTING[export_vars] [RFC2822_DATE]=[Sun, 18 Jun 2017 22:26:34 +0100] to event 2017-06-18 22:26:34.219492 [DEBUG] switch_channel.c:1250 sofia/external/sipp at 185.35.228.51 :5060 EXPORTING[export_vars] [h323-call-origin]=[originate] to event 2017-06-18 22:26:34.219492 [DEBUG] switch_ivr_originate.c:2142 Parsing global variables 2017-06-18 22:26:34.219492 [NOTICE] switch_channel.c:1104 New Channel sofia/external/441554555666 at 185.35.229.30 :5060 [96c1a021-5195-41ce-b903-08b98816d70d] 2017-06-18 22:26:34.219492 [DEBUG] mod_sofia.c:4819 (sofia/external/441554555666 at 185.35.229.30 :5060) State Change CS_NEW -> CS_INIT 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:584 (sofia/external/441554555666 at 185.35.229.30 :5060) Running State Change CS_INIT (Cur 2 Tot 68) 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:627 (sofia/external/441554555666 at 185.35.229.30 :5060) State INIT 2017-06-18 22:26:34.219492 [DEBUG] mod_sofia.c:90 sofia/external/441554555666 at 185.35.229.30 :5060 SOFIA INIT 2017-06-18 22:26:34.219492 [DEBUG] sofia_glue.c:1295 sofia/external/441554555666 at 185.35.229.30 :5060 sending invite version: 1.6.18 git 6e79667 2017-06-12 21:14:49Z 64bit Local SDP: v=0 o=FreeSWITCH 1497789362 1497789363 IN IP4 185.35.228.40 s=FreeSWITCH c=IN IP4 185.35.228.40 t=0 0 m=audio 31832 RTP/AVP 0 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 a=sendrecv 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:40 sofia/external/441554555666 at 185.35.229.30 :5060 Standard INIT 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:48 (sofia/external/441554555666 at 185.35.229.30 :5060) State Change CS_INIT -> CS_ROUTING 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:627 (sofia/external/441554555666 at 185.35.229.30 :5060) State INIT going to sleep 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:584 (sofia/external/441554555666 at 185.35.229.30 :5060) Running State Change CS_ROUTING (Cur 2 Tot 68) 2017-06-18 22:26:34.219492 [DEBUG] sofia.c:7048 Channel sofia/external/441554555666 at 185.35.229.30 :5060 entering state [calling][0] 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:643 (sofia/external/441554555666 at 185.35.229.30 :5060) State ROUTING 2017-06-18 22:26:34.219492 [DEBUG] mod_sofia.c:143 sofia/external/441554555666 at 185.35.229.30 :5060 SOFIA ROUTING 2017-06-18 22:26:34.219492 [DEBUG] switch_ivr_originate.c:67 (sofia/external/441554555666 at 185.35.229.30 :5060) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2017-06-18 22:26:34.219492 [ERR] mod_xml_radius.c:930 Didn't match: 185.35.229.30:5060 == ^185\.35\.229\.30 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:643 (sofia/external/441554555666 at 185.35.229.30 :5060) State ROUTING going to sleep 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:584 (sofia/external/441554555666 at 185.35.229.30 :5060) Running State Change CS_CONSUME_MEDIA (Cur 2 Tot 68) 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:662 (sofia/external/441554555666 at 185.35.229.30 :5060) State CONSUME_MEDIA 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:662 (sofia/external/441554555666 at 185.35.229.30 :5060) State CONSUME_MEDIA going to sleep 2017-06-18 22:26:34.259480 [DEBUG] sofia_glue.c:1295 sofia/external/441554555666 at 185.35.229.30 :5060 sending invite version: 1.6.18 git 6e79667 2017-06-12 21:14:49Z 64bit Local SDP: v=0 o=FreeSWITCH 1497789362 1497789364 IN IP4 185.35.228.40 s=FreeSWITCH c=IN IP4 185.35.228.40 t=0 0 m=audio 31832 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv 2017-06-18 22:26:34.259480 [DEBUG] sofia.c:7048 Channel sofia/external/441554555666 at 185.35.229.30 :5060 entering state [calling][0] 2017-06-18 22:26:34.259480 [DEBUG] sofia.c:7048 Channel sofia/external/441554555666 at 185.35.229.30 :5060 entering state [proceeding][180] 2017-06-18 22:26:34.259480 [NOTICE] sofia.c:7156 Ring-Ready sofia/external/441554555666 at 185.35.229.30 :5060! 2017-06-18 22:26:34.259480 [DEBUG] switch_channel.c:3346 (sofia/external/441554555666 at 185.35.229.30 :5060) Callstate Change DOWN -> RINGING 2017-06-18 22:26:34.259480 [DEBUG] sofia.c:7048 Channel sofia/external/441554555666 at 185.35.229.30 :5060 entering state [completing][200] 2017-06-18 22:26:34.259480 [DEBUG] sofia.c:7058 Remote SDP: v=0 o=user1 53655765 2353687637 IN IP4 185.35.228.48 s=- c=IN IP4 185.35.228.48 t=0 0 m=audio 6000 RTP/AVP 0 a=rtpmap:0 PCMU/8000 2017-06-18 22:26:34.259480 [DEBUG] sofia.c:7048 Channel sofia/external/441554555666 at 185.35.229.30 :5060 entering state [ready][200] 2017-06-18 22:26:34.259480 [DEBUG] switch_core_media.c:4445 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2017-06-18 22:26:34.259480 [DEBUG] switch_core_media.c:4500 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match 2017-06-18 22:26:34.259480 [DEBUG] switch_core_media.c:3057 Set Codec sofia/external/441554555666 at 185.35.229.30 :5060 PCMU/8000 20 ms 160 samples 64000 bits 1 channels 2017-06-18 22:26:34.259480 [DEBUG] switch_core_codec.c:111 sofia/external/441554555666 at 185.35.229.30 :5060 Original read codec set to PCMU:0 2017-06-18 22:26:34.259480 [DEBUG] switch_core_media.c:4770 No 2833 in SDP. Disable 2833 dtmf and switch to INFO 2017-06-18 22:26:34.259480 [DEBUG] switch_core_media.c:6874 AUDIO RTP [sofia/external/441554555666 at 185.35.229.30 :5060] 185.35.228.40 port 31832 -> 185.35.228.48 port 6000 codec: 0 ms: 20 2017-06-18 22:26:34.259480 [DEBUG] switch_rtp.c:4108 Starting timer [soft] 160 bytes per 20ms 2017-06-18 22:26:34.279530 [DEBUG] switch_core_media.c:7205 sofia/external/441554555666 at 185.35.229.30 :5060 Set rtp dtmf delay to 40 2017-06-18 22:26:34.279530 [NOTICE] sofia.c:8182 Channel [sofia/external/441554555666 at 185.35.229.30 :5060] has been answered 2017-06-18 22:26:34.279530 [DEBUG] switch_channel.c:3773 (sofia/external/441554555666 at 185.35.229.30 :5060) Callstate Change RINGING -> ACTIVE 2017-06-18 22:26:34.279530 [NOTICE] mod_sofia.c:2273 Ring-Ready sofia/external/sipp at 185.35.228.51 :5060! 2017-06-18 22:26:34.279530 [DEBUG] sofia.c:7048 Channel sofia/external/sipp at 185.35.228.51 :5060 entering state [early][180] 2017-06-18 22:26:34.279530 [NOTICE] switch_ivr_originate.c:525 Ring Ready sofia/external/sipp at 185.35.228.51 :5060! 2017-06-18 22:26:34.279530 [DEBUG] switch_ivr_originate.c:410 Setting codec string on sofia/external/sipp at 185.35.228.51 :5060 to PCMU at 8000h@20i 2017-06-18 22:26:34.279530 [DEBUG] switch_core_media.c:4445 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] 2017-06-18 22:26:34.279530 [DEBUG] switch_core_media.c:4500 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match 2017-06-18 22:26:34.279530 [DEBUG] switch_core_media.c:3057 Set Codec sofia/external/sipp at 185.35.228.51 :5060 PCMU/8000 20 ms 160 samples 64000 bits 1 channels 2017-06-18 22:26:34.279530 [DEBUG] switch_core_codec.c:111 sofia/external/sipp at 185.35.228.51 :5060 Original read codec set to PCMU:0 2017-06-18 22:26:34.279530 [DEBUG] switch_core_media.c:4770 No 2833 in SDP. Disable 2833 dtmf and switch to INFO 2017-06-18 22:26:34.279530 [DEBUG] switch_core_media.c:6874 AUDIO RTP [sofia/external/sipp at 185.35.228.51 :5060] 185.35.228.40 port 23728 -> 185.35.228.51 port 6000 codec: 0 ms: 20 2017-06-18 22:26:34.279530 [DEBUG] switch_rtp.c:4108 Starting timer [soft] 160 bytes per 20ms 2017-06-18 22:26:34.279530 [DEBUG] switch_core_media.c:7205 sofia/external/sipp at 185.35.228.51 :5060 Set rtp dtmf delay to 40 2017-06-18 22:26:34.279530 [NOTICE] sofia_media.c:92 Pre-Answer sofia/external/sipp at 185.35.228.51 :5060! 2017-06-18 22:26:34.279530 [DEBUG] switch_channel.c:3474 (sofia/external/sipp at 185.35.228.51 :5060) Callstate Change RINGING -> EARLY 2017-06-18 22:26:34.279530 [DEBUG] switch_core_media.c:6857 Audio params are unchanged for sofia/external/sipp at 185.35.228.51 :5060. 2017-06-18 22:26:34.279530 [DEBUG] mod_sofia.c:850 Local SDP sofia/external/sipp at 185.35.228.51 :5060: v=0 o=FreeSWITCH 1497797466 1497797467 IN IP4 185.35.228.40 s=FreeSWITCH c=IN IP4 185.35.228.40 t=0 0 m=audio 23728 RTP/AVP 0 a=rtpmap:0 PCMU/8000 a=ptime:20 a=sendrecv 2017-06-18 22:26:34.279530 [NOTICE] switch_ivr_originate.c:3647 Channel [sofia/external/sipp at 185.35.228.51 :5060] has been answered EXECUTE sofia/external/sipp at 185.35.228.51 :5060 sched_hangup(+ alloted_timeout) 2017-06-18 22:26:34.279530 [DEBUG] sofia.c:7048 Channel sofia/external/sipp at 185.35.228.51 :5060 entering state [completed][200] 2017-06-18 22:26:34.279530 [DEBUG] sofia.c:7048 Channel sofia/external/sipp at 185.35.228.51 :5060 entering state [ready][200] 2017-06-18 22:26:34.279530 [NOTICE] mod_dptools.c:1188 Hangup sofia/external/sipp at 185.35.228.51 :5060 [CS_EXECUTE] [NORMAL_CLEARING] 2017-06-18 22:26:34.279530 [DEBUG] switch_core_session.c:2814 sofia/external/sipp at 185.35.228.51 :5060 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2017-06-18 22:26:34.279530 [DEBUG] switch_channel.c:3773 (sofia/external/sipp at 185.35.228.51 :5060) Callstate Change EARLY -> ACTIVE 2017-06-18 22:26:34.279530 [DEBUG] switch_ivr_originate.c:3647 sofia/external/sipp at 185.35.228.51 :5060 skip receive message [ANSWER_EVENT] (channel is hungup already) 2017-06-18 22:26:34.319525 [DEBUG] switch_ivr_originate.c:3848 Originate Resulted in Error Cause: 487 [ORIGINATOR_CANCEL] 2017-06-18 22:26:34.319525 [NOTICE] switch_ivr_originate.c:3938 Hangup sofia/external/441554555666 at 185.35.229.30 :5060 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL] 2017-06-18 22:26:34.319525 [DEBUG] switch_core_state_machine.c:584 (sofia/external/441554555666 at 185.35.229.30 :5060) Running State Change CS_HANGUP (Cur 2 Tot 68) 2017-06-18 22:26:34.319525 [DEBUG] switch_core_state_machine.c:850 (sofia/external/441554555666 at 185.35.229.30 :5060) Callstate Change ACTIVE -> HANGUP 2017-06-18 22:26:34.319525 [DEBUG] switch_core_state_machine.c:852 (sofia/external/441554555666 at 185.35.229.30 :5060) State HANGUP 2017-06-18 22:26:34.319525 [DEBUG] mod_sofia.c:438 Channel sofia/external/441554555666 at 185.35.229.30 :5060 hanging up, cause: ORIGINATOR_CANCEL 2017-06-18 22:26:34.319525 [INFO] mod_dptools.c:3418 Originate Failed. Cause: ORIGINATOR_CANCEL 2017-06-18 22:26:34.319525 [DEBUG] mod_sofia.c:491 Sending BYE to sofia/external/441554555666 at 185.35.229.30 :5060 2017-06-18 22:26:34.319525 [DEBUG] switch_core_state_machine.c:60 sofia/external/441554555666 at 185.35.229.30 :5060 Standard HANGUP, cause: ORIGINATOR_CANCEL 2017-06-18 22:26:34.319525 [DEBUG] switch_core_state_machine.c:852 (sofia/external/441554555666 at 185.35.229.30 :5060) State HANGUP going to sleep 2017-06-18 22:26:34.319525 [DEBUG] switch_core_state_machine.c:619 (sofia/external/441554555666 at 185.35.229.30 :5060) State Change CS_HANGUP -> CS_REPORTING 2017-06-18 22:26:34.319525 [DEBUG] switch_core_session.c:2814 sofia/external/sipp at 185.35.228.51 :5060 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2017-06-18 22:26:34.319525 [DEBUG] switch_core_state_machine.c:584 (sofia/external/441554555666 at 185.35.229.30 :5060) Running State Change CS_REPORTING (Cur 2 Tot 68) 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:650 (sofia/external/sipp at 185.35.228.51 :5060) State EXECUTE going to sleep 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:584 (sofia/external/sipp at 185.35.228.51 :5060) Running State Change CS_HANGUP (Cur 2 Tot 68) 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:938 (sofia/external/441554555666 at 185.35.229.30 :5060) State REPORTING 2017-06-18 22:26:34.339569 [ERR] mod_xml_radius.c:930 Didn't match: 185.35.229.30 == ^185\.35\.229\.30 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:850 (sofia/external/sipp at 185.35.228.51 :5060) Callstate Change ACTIVE -> HANGUP 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:174 sofia/external/441554555666 at 185.35.229.30 :5060 Standard REPORTING, cause: ORIGINATOR_CANCEL 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:938 (sofia/external/441554555666 at 185.35.229.30 :5060) State REPORTING going to sleep 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:852 (sofia/external/sipp at 185.35.228.51 :5060) State HANGUP 2017-06-18 22:26:34.339569 [DEBUG] mod_sofia.c:438 Channel sofia/external/sipp at 185.35.228.51 :5060 hanging up, cause: NORMAL_CLEARING 2017-06-18 22:26:34.339569 [DEBUG] mod_sofia.c:491 Sending BYE to sofia/external/sipp at 185.35.228.51 :5060 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:60 sofia/external/sipp at 185.35.228.51 :5060 Standard HANGUP, cause: NORMAL_CLEARING 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:852 (sofia/external/sipp at 185.35.228.51 :5060) State HANGUP going to sleep 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:619 (sofia/external/sipp at 185.35.228.51 :5060) State Change CS_HANGUP -> CS_REPORTING 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:584 (sofia/external/sipp at 185.35.228.51 :5060) Running State Change CS_REPORTING (Cur 2 Tot 68) 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:938 (sofia/external/sipp at 185.35.228.51 :5060) State REPORTING 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:610 (sofia/external/441554555666 at 185.35.229.30 :5060) State Change CS_REPORTING -> CS_DESTROY 2017-06-18 22:26:34.339569 [DEBUG] switch_core_session.c:1664 Session 68 (sofia/external/441554555666 at 185.35.229.30 :5060) Locked, Waiting on external entities 2017-06-18 22:26:34.339569 [ERR] mod_xml_radius.c:933 Result of true match: 185.35.228.40 == ^185\.35\.229\.30 2017-06-18 22:26:34.339569 [NOTICE] switch_core_session.c:1682 Session 68 (sofia/external/441554555666 at 185.35.229.30 :5060) Ended 2017-06-18 22:26:34.339569 [NOTICE] switch_core_session.c:1686 Close Channel sofia/external/441554555666 at 185.35.229.30 :5060 [CS_DESTROY] 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:741 (sofia/external/441554555666 at 185.35.229.30 :5060) Running State Change CS_DESTROY (Cur 1 Tot 68) 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:751 (sofia/external/441554555666 at 185.35.229.30 :5060) State DESTROY 2017-06-18 22:26:34.339569 [DEBUG] mod_sofia.c:343 sofia/external/441554555666 at 185.35.229.30 :5060 SOFIA DESTROY 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:181 sofia/external/441554555666 at 185.35.229.30 :5060 Standard DESTROY 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:751 (sofia/external/441554555666 at 185.35.229.30 :5060) State DESTROY going to sleep 2017-06-18 22:26:34.339569 [INFO] mod_xml_radius.c:1044 mod_xml_radius: Accounting Stop success 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:174 sofia/external/sipp at 185.35.228.51 :5060 Standard REPORTING, cause: NORMAL_CLEARING 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:938 (sofia/external/sipp at 185.35.228.51 :5060) State REPORTING going to sleep 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:610 (sofia/external/sipp at 185.35.228.51 :5060) State Change CS_REPORTING -> CS_DESTROY 2017-06-18 22:26:34.339569 [DEBUG] switch_core_session.c:1664 Session 67 (sofia/external/sipp at 185.35.228.51 :5060) Locked, Waiting on external entiti 2017-06-18 22:26:34.339569 [NOTICE] switch_core_session.c:1682 Session 67 (sofia/external/sipp at 185.35.228.51 :5060) Ended 2017-06-18 22:26:34.339569 [NOTICE] switch_core_session.c:1686 Close Channel sofia/external/sipp at 185.35.228.51 :5060 [CS_DESTROY] 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:741 (sofia/external/sipp at 185.35.228.51 :5060) Running State Change CS_DESTROY (Cur 0 Tot 68) 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:751 (sofia/external/sipp at 185.35.228.51 :5060) State DESTROY 2017-06-18 22:26:34.339569 [DEBUG] mod_sofia.c:343 sofia/external/sipp at 185.35.228.51 :5060 SOFIA DESTROY 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:181 sofia/external/sipp at 185.35.228.51 :5060 Standard DESTROY 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:751 (sofia/external/sipp at 185.35.228.51 :5060) State DESTROY going to sleep U 185.35.228.51:5060 -> 185.35.228.40:5080 INVITE sip:441554555666 at 185.35.228.40:5080 SIP/2.0. Via: SIP/2.0/UDP 185.35.228.51:5060;branch=z9hG4bK-27036-1-0. From: sipp >;tag=27036SIPpTag001. To: 441554555666 >. Call-ID: 1-27036 at 185.35.228.51 . CSeq: 1 INVITE. Contact: sip:sipp at 185.35.228.51:5060 . Max-Forwards: 70. Subject: Performance Test. Content-Type: application/sdp. Content-Length: 137. . v=0. o=user1 53655765 2353687637 IN IP4 185.35.228.51. s=-. c=IN IP4 185.35.228.51. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000. # U 185.35.228.40:5080 -> 185.35.228.51:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 185.35.228.51:5060;branch=z9hG4bK-27036-1-0. From: sipp >;tag=27036SIPpTag001. To: 441554555666 >. Call-ID: 1-27036 at 185.35.228.51 . CSeq: 1 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.6.18+git~20170612T211449Z~6e79667c0a~64bit. Content-Length: 0. . # U 185.35.228.40:5080 -> 185.35.228.51:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 185.35.228.51:5060;branch=z9hG4bK-27036-1-0. From: sipp >;tag=27036SIPpTag001. To: 441554555666 >;tag=91vp8601aS4Qp. Call-ID: 1-27036 at 185.35.228.51 . CSeq: 1 INVITE. Contact: >. User-Agent: FreeSWITCH-mod_sofia/1.6.18+git~20170612T211449Z~6e79667c0a~64bit. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY. Supported: timer, path, replaces. Allow-Events: talk, hold, conference, refer. Content-Length: 0. Remote-Party-ID: "Outbound Call" >;party=calling;privacy=off;screen=no. . # U 185.35.228.40:5080 -> 185.35.228.51:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 185.35.228.51:5060;branch=z9hG4bK-27036-1-0. From: sipp >;tag=27036SIPpTag001. To: 441554555666 >;tag=91vp8601aS4Qp. Call-ID: 1-27036 at 185.35.228.51 . CSeq: 1 INVITE. Contact: >. User-Agent: FreeSWITCH-mod_sofia/1.6.18+git~20170612T211449Z~6e79667c0a~64bit. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY. Supported: timer, path, replaces. Allow-Events: talk, hold, conference, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 166. Remote-Party-ID: "Outbound Call" >;party=calling;privacy=off;screen=no. . v=0. o=FreeSWITCH 1497797499 1497797500 IN IP4 185.35.228.40. s=FreeSWITCH. c=IN IP4 185.35.228.40. t=0 0. m=audio 25252 RTP/AVP 0. a=rtpmap:0 PCMU/8000. a=ptime:20. # U 185.35.228.51:5060 -> 185.35.228.40:5080 ACK sip:441554555666 at 185.35.228.40:5080 SIP/2.0. Via: SIP/2.0/UDP 185.35.228.51:5060;branch=z9hG4bK-27036-1-5. From: sipp >;tag=27036SIPpTag001. To: 441554555666 >;tag=91vp8601aS4Qp. Call-ID: 1-27036 at 185.35.228.51 . CSeq: 1 ACK. Contact: sip:sipp at 185.35.228.51:5060 . Max-Forwards: 70. Subject: Performance Test. Content-Length: 0. . # U 185.35.228.40:5080 -> 185.35.228.51:5060 BYE sip:sipp at 185.35.228.51:5060 SIP/2.0. Via: SIP/2.0/UDP 185.35.228.40:5080;rport;branch=z9hG4bKUBe0HvDjHF1XN. Max-Forwards: 70. From: 441554555666 >;tag=91vp8601aS4Qp. To: sipp >;tag=27036SIPpTag001. Call-ID: 1-27036 at 185.35.228.51 . CSeq: 108575311 BYE. User-Agent: FreeSWITCH-mod_sofia/1.6.18+git~20170612T211449Z~6e79667c0a~64bit. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY. Supported: timer, path, replaces. Reason: Q.850;cause=16;text="NORMAL_CLEARING". Content-Length: 0. . # U 185.35.228.51:5060 -> 185.35.228.40:5080 SIP/2.0 200 OK. Via: SIP/2.0/UDP 185.35.228.40:5080;rport;branch=z9hG4bKUBe0HvDjHF1XN. From: 441554555666 >;tag=91vp8601aS4Qp. To: sipp >;tag=27036SIPpTag001. Call-ID: 1-27036 at 185.35.228.51 . CSeq: 108575311 BYE. Contact: >. Content-Length: 0. NGREP of SIP messages from FS to terminator U 185.35.228.40:5080 -> 185.35.228.48:5060 INVITE sip:441554555666 at 185.35.228.48:5060 SIP/2.0. Via: SIP/2.0/UDP 185.35.228.40:5080;rport;branch=z9hG4bKXX0HNjFSB1D3c. Max-Forwards: 69. From: "sipp" ;tag=eFUjHeNKv5KNg. To: . Call-ID: c755e7c8-cf13-1235-25a9-363165383663. CSeq: 108575420 INVITE. Contact: . User-Agent: FreeSWITCH-mod_sofia/1.6.18+git~20170612T211449Z~6e79667c0a~64bit. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY. Supported: timer, path, replaces. Allow-Events: talk, hold, conference, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 222. X-FS-Support: update_display,send_info. Remote-Party-ID: "sipp" ;party=calling;screen=yes;privacy=off. . v=0. o=FreeSWITCH 1497801741 1497801743 IN IP4 185.35.228.40. s=FreeSWITCH. c=IN IP4 185.35.228.40. t=0 0. m=audio 21228 RTP/AVP 0 101. a=rtpmap:0 PCMU/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. # U 185.35.228.48:5060 -> 185.35.228.40:5080 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 185.35.228.40:5080;rport;branch=z9hG4bKXX0HNjFSB1D3c. From: "sipp" ;tag=eFUjHeNKv5KNg. To: ;tag=31480SIPpTag018. Call-ID: c755e7c8-cf13-1235-25a9-363165383663. CSeq: 108575420 INVITE. Contact: . Content-Length: 0. . # U 185.35.228.48:5060 -> 185.35.228.40:5080 SIP/2.0 200 OK. Via: SIP/2.0/UDP 185.35.228.40:5080;rport;branch=z9hG4bKXX0HNjFSB1D3c. From: "sipp" ;tag=eFUjHeNKv5KNg. To: ;tag=31480SIPpTag018. Call-ID: c755e7c8-cf13-1235-25a9-363165383663. CSeq: 108575420 INVITE. Contact: . Content-Type: application/sdp. Content-Length: 137. . v=0. o=user1 53655765 2353687637 IN IP4 185.35.228.48. s=-. c=IN IP4 185.35.228.48. t=0 0. m=audio 6000 RTP/AVP 0. a=rtpmap:0 PCMU/8000. # U 185.35.228.40:5080 -> 185.35.228.48:5060 ACK sip:185.35.228.48:5060;transport=UDP SIP/2.0. Via: SIP/2.0/UDP 185.35.228.40:5080;rport;branch=z9hG4bKy6SaQD0v893Nr. Max-Forwards: 70. From: "sipp" ;tag=eFUjHeNKv5KNg. To: ;tag=31480SIPpTag018. Call-ID: c755e7c8-cf13-1235-25a9-363165383663. CSeq: 108575420 ACK. Contact: . Content-Length: 0. . # U 185.35.228.40:5080 -> 185.35.228.48:5060 BYE sip:185.35.228.48:5060;transport=UDP SIP/2.0. Via: SIP/2.0/UDP 185.35.228.40:5080;rport;branch=z9hG4bKZFK3r8g05jt8K. Max-Forwards: 70. From: "sipp" ;tag=eFUjHeNKv5KNg. To: ;tag=31480SIPpTag018. Call-ID: c755e7c8-cf13-1235-25a9-363165383663. CSeq: 108575421 BYE. User-Agent: FreeSWITCH-mod_sofia/1.6.18+git~20170612T211449Z~6e79667c0a~64bit. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY. Supported: timer, path, replaces. Reason: SIP;cause=487;text="ORIGINATOR_CANCEL". Content-Length: 0. . # U 185.35.228.48:5060 -> 185.35.228.40:5080 SIP/2.0 200 OK. Via: SIP/2.0/UDP 185.35.228.40:5080;rport;branch=z9hG4bKZFK3r8g05jt8K. From: "sipp" ;tag=eFUjHeNKv5KNg. To: ;tag=31480SIPpTag018. Call-ID: c755e7c8-cf13-1235-25a9-363165383663. CSeq: 108575421 BYE. Contact: . Content-Length: 0. . -------------- next part -------------- An HTML attachment was scrubbed... URL: From colin.morelli at gmail.com Sun Jun 18 23:08:18 2017 From: colin.morelli at gmail.com (Colin Morelli) Date: Sun, 18 Jun 2017 19:08:18 -0400 Subject: [Freeswitch-users] Call Dropping In-Reply-To: <222F1F16-373C-484A-B290-200CB13F2CAD@tm.net.uk> References: <222F1F16-373C-484A-B290-200CB13F2CAD@tm.net.uk> Message-ID: You've got an execute_on_answer of sched_hangup(+${h323-credit-time} alloted_timeout) Immediately after your call is answered: 2017-06-18 22:26:34.279530 [NOTICE] switch_ivr_originate.c:3647 Channel [ sofia/external/sipp at 185.35.228.51:5060] has been answered EXECUTE sofia/external/sipp at 185.35.228.51:5060 sched_hangup(+ alloted_timeout) It would seem that h323-credit-time is not being set, which is causing sched_hangup to immediately hangup the call on answer. On Sun, Jun 18, 2017 at 6:52 PM, Joseph Waite wrote: > Hi Guys > > Using FreeSwitch with Radius linked to JeraSoft VCS billing system. > > I am sending a Call from a SIPP originator, through the FreeSwitch box and > back out to another SIPP terminator scenario. > The call goes through ok, everything happens as it should, however the > call immediately drops, I have done egrep’s of both sides of the call and > the BYE is defiantly coming from Freeswitch for some reason but I cannot > work out why. Anyone any ideas? > I am attaching the FreeSwitch logs plus the egrep’s > If I register zipper on my laptop to FS and make a call works fine. > > 2017-06-18 22:26:34.199519 [NOTICE] switch_channel.c:1104 New Channel > sofia/external/sipp at 185.35.228.51:5060 [15021010-8f64-439f-8dbb- > 1afe090c44a5] > 2017-06-18 22:26:34.199519 [DEBUG] switch_core_state_machine.c:584 ( > sofia/external/sipp at 185.35.228.51:5060) Running State Change CS_NEW (Cur > 1 Tot 67) > 2017-06-18 22:26:34.199519 [DEBUG] sofia.c:9837 sofia/external/ > sipp at 185.35.228.51:5060 receiving invite from 185.35.228.51:5060 version: > 1.6.18 git 6e79667 2017-06-12 21:14:49Z 64bit > 2017-06-18 22:26:34.219492 [DEBUG] sofia.c:7048 Channel > sofia/external/sipp at 185.35.228.51:5060 entering state [received][100] > 2017-06-18 22:26:34.219492 [DEBUG] sofia.c:7058 Remote SDP: > v=0 > o=user1 53655765 2353687637 IN IP4 185.35.228.51 > s=- > c=IN IP4 185.35.228.51 > t=0 0 > m=audio 6000 RTP/AVP 0 > a=rtpmap:0 PCMU/8000 > > 2017-06-18 22:26:34.219492 [DEBUG] sofia.c:7450 ( > sofia/external/sipp at 185.35.228.51:5060) State Change CS_NEW -> CS_INIT > 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:603 ( > sofia/external/sipp at 185.35.228.51:5060) State NEW > 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:584 ( > sofia/external/sipp at 185.35.228.51:5060) Running State Change CS_INIT (Cur > 1 Tot 67) > 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:627 ( > sofia/external/sipp at 185.35.228.51:5060) State INIT > 2017-06-18 22:26:34.219492 [DEBUG] mod_sofia.c:90 sofia/external/ > sipp at 185.35.228.51:5060 SOFIA INIT > 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:40 > sofia/external/sipp at 185.35.228.51:5060 Standard INIT > 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:48 ( > sofia/external/sipp at 185.35.228.51:5060) State Change CS_INIT -> CS_ROUTING > 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:627 ( > sofia/external/sipp at 185.35.228.51:5060) State INIT going to sleep > 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:584 ( > sofia/external/sipp at 185.35.228.51:5060) Running State Change CS_ROUTING > (Cur 1 Tot 67) > 2017-06-18 22:26:34.219492 [DEBUG] switch_channel.c:2249 ( > sofia/external/sipp at 185.35.228.51:5060) Callstate Change DOWN -> RINGING > 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:643 ( > sofia/external/sipp at 185.35.228.51:5060) State ROUTING > 2017-06-18 22:26:34.219492 [DEBUG] mod_sofia.c:143 sofia/ > external/sipp at 185.35.228.51:5060 SOFIA ROUTING > 2017-06-18 22:26:34.219492 [ERR] mod_xml_radius.c:933 Result of true > match: 185.35.228.40 == ^185\.35\.229\.30 > 2017-06-18 22:26:34.219492 [INFO] mod_xml_radius.c:986 mod_xml_radius: > Accounting Start success > 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:236 > sofia/external/sipp at 185.35.228.51:5060 Standard ROUTING > 2017-06-18 22:26:34.219492 [INFO] mod_dialplan_xml.c:637 Processing sipp > ->441554555666 in context public > Dialplan: sofia/external/sipp at 185.35.228.51:5060 parsing [public->unloop] > continue=false > Dialplan: sofia/external/sipp at 185.35.228.51:5060 Regex (PASS) [unloop] > ${unroll_loops}(true) =~ /^true$/ break=on-false > Dialplan: sofia/external/sipp at 185.35.228.51:5060 Regex (FAIL) [unloop] > ${sip_looped_call}() =~ /^true$/ break=on-false > Dialplan: sofia/external/sipp at 185.35.228.51:5060 parsing > [public->outside_call] continue=true > Dialplan: sofia/external/sipp at 185.35.228.51:5060 Absolute Condition > [outside_call] > Dialplan: sofia/external/sipp at 185.35.228.51:5060 Action > set(outside_call=true) > Dialplan: sofia/external/sipp at 185.35.228.51:5060 Action > export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) > Dialplan: sofia/external/sipp at 185.35.228.51:5060 parsing > [public->call_debug] continue=true > Dialplan: sofia/external/sipp at 185.35.228.51:5060 Regex (FAIL) > [call_debug] ${call_debug}(false) =~ /^true$/ break=never > Dialplan: sofia/external/sipp at 185.35.228.51:5060 parsing > [public->rejections] continue=false > Dialplan: sofia/external/sipp at 185.35.228.51:5060 Regex (FAIL) > [rejections] ${radius_auth_result}() =~ /2/ break=on-false > Dialplan: sofia/external/sipp at 185.35.228.51:5060 parsing > [public->timedouts] continue=false > Dialplan: sofia/external/sipp at 185.35.228.51:5060 Regex (FAIL) [timedouts] > ${radius_auth_result}() =~ /1/ break=on-false > Dialplan: sofia/external/sipp at 185.35.228.51:5060 parsing > [public->JeraSoft VCS Routing] continue=false > Dialplan: sofia/external/sipp at 185.35.228.51:5060 Regex (PASS) [JeraSoft > VCS Routing] destination_number(441554555666) =~ /^(.+)$/ break=on-false > Dialplan: sofia/external/sipp at 185.35.228.51:5060 Action > export(nolocal:h323-call-origin=originate) > Dialplan: sofia/external/sipp at 185.35.228.51:5060 Action > set(sip_h_X-accountcode=${accountcode}) > Dialplan: sofia/external/sipp at 185.35.228.51:5060 Action > set(call_direction=outbound) > Dialplan: sofia/external/sipp at 185.35.228.51:5060 Action > set(hangup_after_bridge=true) > Dialplan: sofia/external/sipp at 185.35.228.51:5060 Action > set(continue_on_fail=true) > Dialplan: sofia/external/sipp at 185.35.228.51:5060 Action > set(inherit_codec=true) > Dialplan: sofia/external/sipp at 185.35.228.51:5060 Action > set(call_timeout=20) > Dialplan: sofia/external/sipp at 185.35.228.51:5060 Action > set(fail_on_single_reject=USER_BUSY) > Dialplan: sofia/external/sipp at 185.35.228.51:5060 Action > set(origination_caller_id_name=${sip_req_user}) > Dialplan: sofia/external/sipp at 185.35.228.51:5060 Action > set(origination_caller_id_number=${sip_from_user}) > Dialplan: sofia/external/sipp at 185.35.228.51:5060 Action > set(execute_on_answer=sched_hangup +${h323-credit-time} alloted_timeout) > Dialplan: sofia/external/sipp at 185.35.228.51:5060 Action > bridge({sip_invite_from_uri=sip:${sip_from_user}@${sip_ > network_ip}}sofia/external/${destination_number}@185.35.229.30:5060 > ) > Dialplan: sofia/external/sipp at 185.35.228.51:5060 Action > hangup(${bridge_hangup_cause}) > 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:286 ( > sofia/external/sipp at 185.35.228.51:5060) State Change CS_ROUTING -> > CS_EXECUTE > 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:643 ( > sofia/external/sipp at 185.35.228.51:5060) State ROUTING going to sleep > 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:584 ( > sofia/external/sipp at 185.35.228.51:5060) Running State Change CS_EXECUTE > (Cur 1 Tot 67) > 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:650 ( > sofia/external/sipp at 185.35.228.51:5060) State EXECUTE > 2017-06-18 22:26:34.219492 [DEBUG] mod_sofia.c:198 sofia/ > external/sipp at 185.35.228.51:5060 SOFIA EXECUTE > 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:328 > sofia/external/sipp at 185.35.228.51:5060 Standard EXECUTE > EXECUTE sofia/external/sipp at 185.35.228.51:5060 set(outside_call=true) > 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET > sofia/external/sipp at 185.35.228.51:5060 [outside_call]=[true] > EXECUTE sofia/external/sipp at 185.35.228.51:5060 export(RFC2822_DATE=Sun, > 18 Jun 2017 22:26:34 +0100) > 2017-06-18 22:26:34.219492 [DEBUG] switch_channel.c:1296 EXPORT > (export_vars) [RFC2822_DATE]=[Sun, 18 Jun 2017 22:26:34 +0100] > EXECUTE sofia/external/sipp at 185.35.228.51:5060 export(nolocal:h323-call- > origin=originate) > 2017-06-18 22:26:34.219492 [DEBUG] switch_channel.c:1296 EXPORT > (export_vars) (REMOTE ONLY) [h323-call-origin]=[originate] > EXECUTE sofia/external/sipp at 185.35.228.51:5060 set(sip_h_X-accountcode=) > 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET > sofia/external/sipp at 185.35.228.51:5060 [sip_h_X-accountcode]=[UNDEF] > EXECUTE sofia/external/sipp at 185.35.228.51:5060 > set(call_direction=outbound) > 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET > sofia/external/sipp at 185.35.228.51:5060 [call_direction]=[outbound] > EXECUTE sofia/external/sipp at 185.35.228.51:5060 > set(hangup_after_bridge=true) > 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET > sofia/external/sipp at 185.35.228.51:5060 [hangup_after_bridge]=[true] > EXECUTE sofia/external/sipp at 185.35.228.51:5060 set(continue_on_fail=true) > 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET > sofia/external/sipp at 185.35.228.51:5060 [continue_on_fail]=[true] > EXECUTE sofia/external/sipp at 185.35.228.51:5060 set(inherit_codec=true) > 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET > sofia/external/sipp at 185.35.228.51:5060 [inherit_codec]=[true] > EXECUTE sofia/external/sipp at 185.35.228.51:5060 set(call_timeout=20) > 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET > sofia/external/sipp at 185.35.228.51:5060 [call_timeout]=[20] > EXECUTE sofia/external/sipp at 185.35.228.51:5060 set(fail_on_single_reject= > USER_BUSY) > 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET > sofia/external/sipp at 185.35.228.51:5060 [fail_on_single_reject]=[USER_BUSY] > EXECUTE sofia/external/sipp at 185.35.228.51:5060 set(origination_caller_id_ > name=441554555666) > 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET > sofia/external/sipp at 185.35.228.51:5060 [origination_caller_id_name]=[ > 441554555666] > EXECUTE sofia/external/sipp at 185.35.228.51:5060 set(origination_caller_id_ > number=sipp) > 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET > sofia/external/sipp at 185.35.228.51:5060 [origination_caller_id_number] > =[sipp] > EXECUTE sofia/external/sipp at 185.35.228.51:5060 > set(execute_on_answer=sched_hangup + alloted_timeout) > 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET > sofia/external/sipp at 185.35.228.51:5060 [execute_on_answer]=[sched_hangup > + alloted_timeout] > EXECUTE sofia/external/sipp at 185.35.228.51:5060 > bridge({sip_invite_from_uri=sip:sipp at 185.35.228.51}sofia/ > external/441554555666 at 185.35.229.30:5060) > 2017-06-18 22:26:34.219492 [DEBUG] switch_channel.c:1250 sofia/ > external/sipp at 185.35.228.51:5060 EXPORTING[export_vars] > [RFC2822_DATE]=[Sun, 18 Jun 2017 22:26:34 +0100] to event > 2017-06-18 22:26:34.219492 [DEBUG] switch_channel.c:1250 sofia/ > external/sipp at 185.35.228.51:5060 EXPORTING[export_vars] > [h323-call-origin]=[originate] to event > 2017-06-18 22:26:34.219492 [DEBUG] switch_ivr_originate.c:2142 Parsing > global variables > 2017-06-18 22:26:34.219492 [NOTICE] switch_channel.c:1104 New Channel > sofia/external/441554555666 at 185.35.229.30:5060 [96c1a021-5195-41ce-b903- > 08b98816d70d] > 2017-06-18 22:26:34.219492 [DEBUG] mod_sofia.c:4819 ( > sofia/external/441554555666 at 185.35.229.30:5060) State Change CS_NEW -> > CS_INIT > 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:584 ( > sofia/external/441554555666 at 185.35.229.30:5060) Running State Change > CS_INIT (Cur 2 Tot 68) > 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:627 ( > sofia/external/441554555666 at 185.35.229.30:5060) State INIT > 2017-06-18 22:26:34.219492 [DEBUG] mod_sofia.c:90 sofia/external/ > 441554555666 at 185.35.229.30:5060 SOFIA INIT > 2017-06-18 22:26:34.219492 [DEBUG] sofia_glue.c:1295 sofia/ > external/441554555666 at 185.35.229.30:5060 sending invite version: 1.6.18 > git 6e79667 2017-06-12 21:14:49Z 64bit > Local SDP: > v=0 > o=FreeSWITCH 1497789362 1497789363 IN IP4 185.35.228.40 > s=FreeSWITCH > c=IN IP4 185.35.228.40 > t=0 0 > m=audio 31832 RTP/AVP 0 101 13 > a=rtpmap:0 PCMU/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:13 CN/8000 > a=ptime:20 > a=sendrecv > > 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:40 > sofia/external/441554555666 at 185.35.229.30:5060 Standard INIT > 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:48 ( > sofia/external/441554555666 at 185.35.229.30:5060) State Change CS_INIT -> > CS_ROUTING > 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:627 ( > sofia/external/441554555666 at 185.35.229.30:5060) State INIT going to sleep > 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:584 ( > sofia/external/441554555666 at 185.35.229.30:5060) Running State Change > CS_ROUTING (Cur 2 Tot 68) > 2017-06-18 22:26:34.219492 [DEBUG] sofia.c:7048 Channel sofia/external/ > 441554555666 at 185.35.229.30:5060 entering state [calling][0] > 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:643 ( > sofia/external/441554555666 at 185.35.229.30:5060) State ROUTING > 2017-06-18 22:26:34.219492 [DEBUG] mod_sofia.c:143 sofia/ > external/441554555666 at 185.35.229.30:5060 SOFIA ROUTING > 2017-06-18 22:26:34.219492 [DEBUG] switch_ivr_originate.c:67 ( > sofia/external/441554555666 at 185.35.229.30:5060) State Change CS_ROUTING > -> CS_CONSUME_MEDIA > 2017-06-18 22:26:34.219492 [ERR] mod_xml_radius.c:930 Didn't match: > 185.35.229.30:5060 == ^185\.35\.229\.30 > 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:643 ( > sofia/external/441554555666 at 185.35.229.30:5060) State ROUTING going to > sleep > 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:584 ( > sofia/external/441554555666 at 185.35.229.30:5060) Running State Change > CS_CONSUME_MEDIA (Cur 2 Tot 68) > 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:662 ( > sofia/external/441554555666 at 185.35.229.30:5060) State CONSUME_MEDIA > 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:662 ( > sofia/external/441554555666 at 185.35.229.30:5060) State CONSUME_MEDIA going > to sleep > 2017-06-18 22:26:34.259480 [DEBUG] sofia_glue.c:1295 sofia/ > external/441554555666 at 185.35.229.30:5060 sending invite version: 1.6.18 > git 6e79667 2017-06-12 21:14:49Z 64bit > Local SDP: > v=0 > o=FreeSWITCH 1497789362 1497789364 IN IP4 185.35.228.40 > s=FreeSWITCH > c=IN IP4 185.35.228.40 > t=0 0 > m=audio 31832 RTP/AVP 0 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > a=sendrecv > > 2017-06-18 22:26:34.259480 [DEBUG] sofia.c:7048 Channel sofia/external/ > 441554555666 at 185.35.229.30:5060 entering state [calling][0] > 2017-06-18 22:26:34.259480 [DEBUG] sofia.c:7048 Channel sofia/external/ > 441554555666 at 185.35.229.30:5060 entering state [proceeding][180] > 2017-06-18 22:26:34.259480 [NOTICE] sofia.c:7156 Ring-Ready > sofia/external/441554555666 at 185.35.229.30:5060! > 2017-06-18 22:26:34.259480 [DEBUG] switch_channel.c:3346 ( > sofia/external/441554555666 at 185.35.229.30:5060) Callstate Change DOWN -> > RINGING > 2017-06-18 22:26:34.259480 [DEBUG] sofia.c:7048 Channel sofia/external/ > 441554555666 at 185.35.229.30:5060 entering state [completing][200] > 2017-06-18 22:26:34.259480 [DEBUG] sofia.c:7058 Remote SDP: > v=0 > o=user1 53655765 2353687637 IN IP4 185.35.228.48 > s=- > c=IN IP4 185.35.228.48 > t=0 0 > m=audio 6000 RTP/AVP 0 > a=rtpmap:0 PCMU/8000 > > 2017-06-18 22:26:34.259480 [DEBUG] sofia.c:7048 Channel sofia/external/ > 441554555666 at 185.35.229.30:5060 entering state [ready][200] > 2017-06-18 22:26:34.259480 [DEBUG] switch_core_media.c:4445 Audio Codec > Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] > 2017-06-18 22:26:34.259480 [DEBUG] switch_core_media.c:4500 Audio Codec > Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match > 2017-06-18 22:26:34.259480 [DEBUG] switch_core_media.c:3057 Set Codec > sofia/external/441554555666 at 185.35.229.30:5060 PCMU/8000 20 ms 160 > samples 64000 bits 1 channels > 2017-06-18 22:26:34.259480 [DEBUG] switch_core_codec.c:111 sofia/ > external/441554555666 at 185.35.229.30:5060 Original read codec set to PCMU:0 > 2017-06-18 22:26:34.259480 [DEBUG] switch_core_media.c:4770 No 2833 in > SDP. Disable 2833 dtmf and switch to INFO > 2017-06-18 22:26:34.259480 [DEBUG] switch_core_media.c:6874 AUDIO RTP [ > sofia/external/441554555666 at 185.35.229.30:5060] 185.35.228.40 port 31832 > -> 185.35.228.48 port 6000 codec: 0 ms: 20 > 2017-06-18 22:26:34.259480 [DEBUG] switch_rtp.c:4108 Starting timer [soft] > 160 bytes per 20ms > 2017-06-18 22:26:34.279530 [DEBUG] switch_core_media.c:7205 sofia > /external/441554555666 at 185.35.229.30:5060 Set rtp dtmf delay to 40 > 2017-06-18 22:26:34.279530 [NOTICE] sofia.c:8182 Channel [ > sofia/external/441554555666 at 185.35.229.30:5060] has been answered > 2017-06-18 22:26:34.279530 [DEBUG] switch_channel.c:3773 ( > sofia/external/441554555666 at 185.35.229.30:5060) Callstate Change RINGING > -> ACTIVE > 2017-06-18 22:26:34.279530 [NOTICE] mod_sofia.c:2273 Ring-Ready > sofia/external/sipp at 185.35.228.51:5060! > 2017-06-18 22:26:34.279530 [DEBUG] sofia.c:7048 Channel > sofia/external/sipp at 185.35.228.51:5060 entering state [early][180] > 2017-06-18 22:26:34.279530 [NOTICE] switch_ivr_originate.c:525 Ring Ready > sofia/external/sipp at 185.35.228.51:5060! > 2017-06-18 22:26:34.279530 [DEBUG] switch_ivr_originate.c:410 Setting > codec string on sofia/external/sipp at 185.35.228.51:5060 to PCMU at 8000h@20i > 2017-06-18 22:26:34.279530 [DEBUG] switch_core_media.c:4445 Audio Codec > Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] > 2017-06-18 22:26:34.279530 [DEBUG] switch_core_media.c:4500 Audio Codec > Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match > 2017-06-18 22:26:34.279530 [DEBUG] switch_core_media.c:3057 Set Codec > sofia/external/sipp at 185.35.228.51:5060 PCMU/8000 20 ms 160 samples 64000 > bits 1 channels > 2017-06-18 22:26:34.279530 [DEBUG] switch_core_codec.c:111 sofia/ > external/sipp at 185.35.228.51:5060 Original read codec set to PCMU:0 > 2017-06-18 22:26:34.279530 [DEBUG] switch_core_media.c:4770 No 2833 in > SDP. Disable 2833 dtmf and switch to INFO > 2017-06-18 22:26:34.279530 [DEBUG] switch_core_media.c:6874 AUDIO RTP [ > sofia/external/sipp at 185.35.228.51:5060] 185.35.228.40 port 23728 -> > 185.35.228.51 port 6000 codec: 0 ms: 20 > 2017-06-18 22:26:34.279530 [DEBUG] switch_rtp.c:4108 Starting timer [soft] > 160 bytes per 20ms > 2017-06-18 22:26:34.279530 [DEBUG] switch_core_media.c:7205 sofia > /external/sipp at 185.35.228.51:5060 Set rtp dtmf delay to 40 > 2017-06-18 22:26:34.279530 [NOTICE] sofia_media.c:92 Pre-Answer > sofia/external/sipp at 185.35.228.51:5060! > 2017-06-18 22:26:34.279530 [DEBUG] switch_channel.c:3474 ( > sofia/external/sipp at 185.35.228.51:5060) Callstate Change RINGING -> EARLY > 2017-06-18 22:26:34.279530 [DEBUG] switch_core_media.c:6857 Audio params > are unchanged for sofia/external/sipp at 185.35.228.51:5060. > 2017-06-18 22:26:34.279530 [DEBUG] mod_sofia.c:850 Local SDP > sofia/external/sipp at 185.35.228.51:5060: > v=0 > o=FreeSWITCH 1497797466 1497797467 IN IP4 185.35.228.40 > s=FreeSWITCH > c=IN IP4 185.35.228.40 > t=0 0 > m=audio 23728 RTP/AVP 0 > a=rtpmap:0 PCMU/8000 > a=ptime:20 > a=sendrecv > > 2017-06-18 22:26:34.279530 [NOTICE] switch_ivr_originate.c:3647 Channel [ > sofia/external/sipp at 185.35.228.51:5060] has been answered > EXECUTE sofia/external/sipp at 185.35.228.51:5060 sched_hangup(+ > alloted_timeout) > 2017-06-18 22:26:34.279530 [DEBUG] sofia.c:7048 Channel > sofia/external/sipp at 185.35.228.51:5060 entering state [completed][200] > 2017-06-18 22:26:34.279530 [DEBUG] sofia.c:7048 Channel > sofia/external/sipp at 185.35.228.51:5060 entering state [ready][200] > 2017-06-18 22:26:34.279530 [NOTICE] mod_dptools.c:1188 Hangup > sofia/external/sipp at 185.35.228.51:5060 [CS_EXECUTE] [NORMAL_CLEARING] > 2017-06-18 22:26:34.279530 [DEBUG] switch_core_session.c:2814 sof > ia/external/sipp at 185.35.228.51:5060 skip receive message > [APPLICATION_EXEC_COMPLETE] (channel is hungup already) > 2017-06-18 22:26:34.279530 [DEBUG] switch_channel.c:3773 ( > sofia/external/sipp at 185.35.228.51:5060) Callstate Change EARLY -> ACTIVE > 2017-06-18 22:26:34.279530 [DEBUG] switch_ivr_originate.c:3647 so > fia/external/sipp at 185.35.228.51:5060 skip receive message [ANSWER_EVENT] > (channel is hungup already) > 2017-06-18 22:26:34.319525 [DEBUG] switch_ivr_originate.c:3848 Originate > Resulted in Error Cause: 487 [ORIGINATOR_CANCEL] > 2017-06-18 22:26:34.319525 [NOTICE] switch_ivr_originate.c:3938 Hangup > sofia/external/441554555666 at 185.35.229.30:5060 [CS_CONSUME_MEDIA] > [ORIGINATOR_CANCEL] > 2017-06-18 22:26:34.319525 [DEBUG] switch_core_state_machine.c:584 ( > sofia/external/441554555666 at 185.35.229.30:5060) Running State Change > CS_HANGUP (Cur 2 Tot 68) > 2017-06-18 22:26:34.319525 [DEBUG] switch_core_state_machine.c:850 ( > sofia/external/441554555666 at 185.35.229.30:5060) Callstate Change ACTIVE > -> HANGUP > 2017-06-18 22:26:34.319525 [DEBUG] switch_core_state_machine.c:852 ( > sofia/external/441554555666 at 185.35.229.30:5060) State HANGUP > 2017-06-18 22:26:34.319525 [DEBUG] mod_sofia.c:438 Channel sofia/external/ > 441554555666 at 185.35.229.30:5060 hanging up, cause: ORIGINATOR_CANCEL > 2017-06-18 22:26:34.319525 [INFO] mod_dptools.c:3418 Originate Failed. > Cause: ORIGINATOR_CANCEL > 2017-06-18 22:26:34.319525 [DEBUG] mod_sofia.c:491 Sending BYE to > sofia/external/441554555666 at 185.35.229.30:5060 > 2017-06-18 22:26:34.319525 [DEBUG] switch_core_state_machine.c:60 > sofia/external/441554555666 at 185.35.229.30:5060 Standard HANGUP, cause: > ORIGINATOR_CANCEL > 2017-06-18 22:26:34.319525 [DEBUG] switch_core_state_machine.c:852 ( > sofia/external/441554555666 at 185.35.229.30:5060) State HANGUP going to > sleep > 2017-06-18 22:26:34.319525 [DEBUG] switch_core_state_machine.c:619 ( > sofia/external/441554555666 at 185.35.229.30:5060) State Change CS_HANGUP -> > CS_REPORTING > 2017-06-18 22:26:34.319525 [DEBUG] switch_core_session.c:2814 sof > ia/external/sipp at 185.35.228.51:5060 skip receive message > [APPLICATION_EXEC_COMPLETE] (channel is hungup already) > 2017-06-18 22:26:34.319525 [DEBUG] switch_core_state_machine.c:584 ( > sofia/external/441554555666 at 185.35.229.30:5060) Running State Change > CS_REPORTING (Cur 2 Tot 68) > 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:650 ( > sofia/external/sipp at 185.35.228.51:5060) State EXECUTE going to sleep > 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:584 ( > sofia/external/sipp at 185.35.228.51:5060) Running State Change CS_HANGUP > (Cur 2 Tot 68) > 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:938 ( > sofia/external/441554555666 at 185.35.229.30:5060) State REPORTING > 2017-06-18 22:26:34.339569 [ERR] mod_xml_radius.c:930 Didn't match: > 185.35.229.30 == ^185\.35\.229\.30 > 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:850 ( > sofia/external/sipp at 185.35.228.51:5060) Callstate Change ACTIVE -> HANGUP > 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:174 > sofia/external/441554555666 at 185.35.229.30:5060 Standard REPORTING, cause: > ORIGINATOR_CANCEL > 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:938 ( > sofia/external/441554555666 at 185.35.229.30:5060) State REPORTING going to > sleep > 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:852 ( > sofia/external/sipp at 185.35.228.51:5060) State HANGUP > 2017-06-18 22:26:34.339569 [DEBUG] mod_sofia.c:438 Channel > sofia/external/sipp at 185.35.228.51:5060 hanging up, cause: NORMAL_CLEARING > 2017-06-18 22:26:34.339569 [DEBUG] mod_sofia.c:491 Sending BYE to > sofia/external/sipp at 185.35.228.51:5060 > 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:60 > sofia/external/sipp at 185.35.228.51:5060 Standard HANGUP, cause: > NORMAL_CLEARING > 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:852 ( > sofia/external/sipp at 185.35.228.51:5060) State HANGUP going to sleep > 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:619 ( > sofia/external/sipp at 185.35.228.51:5060) State Change CS_HANGUP -> > CS_REPORTING > 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:584 ( > sofia/external/sipp at 185.35.228.51:5060) Running State Change CS_REPORTING > (Cur 2 Tot 68) > 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:938 ( > sofia/external/sipp at 185.35.228.51:5060) State REPORTING > 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:610 ( > sofia/external/441554555666 at 185.35.229.30:5060) State Change CS_REPORTING > -> CS_DESTROY > 2017-06-18 22:26:34.339569 [DEBUG] switch_core_session.c:1664 Session 68 ( > sofia/external/441554555666 at 185.35.229.30:5060) Locked, Waiting on > external entities > 2017-06-18 22:26:34.339569 [ERR] mod_xml_radius.c:933 Result of true > match: 185.35.228.40 == ^185\.35\.229\.30 > 2017-06-18 22:26:34.339569 [NOTICE] switch_core_session.c:1682 Session 68 ( > sofia/external/441554555666 at 185.35.229.30:5060) Ended > 2017-06-18 22:26:34.339569 [NOTICE] switch_core_session.c:1686 Close > Channel sofia/external/441554555666 at 185.35.229.30:5060 [CS_DESTROY] > 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:741 ( > sofia/external/441554555666 at 185.35.229.30:5060) Running State Change > CS_DESTROY (Cur 1 Tot 68) > 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:751 ( > sofia/external/441554555666 at 185.35.229.30:5060) State DESTROY > 2017-06-18 22:26:34.339569 [DEBUG] mod_sofia.c:343 sofia/ > external/441554555666 at 185.35.229.30:5060 SOFIA DESTROY > 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:181 > sofia/external/441554555666 at 185.35.229.30:5060 Standard DESTROY > 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:751 ( > sofia/external/441554555666 at 185.35.229.30:5060) State DESTROY going to > sleep > 2017-06-18 22:26:34.339569 [INFO] mod_xml_radius.c:1044 mod_xml_radius: > Accounting Stop success > 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:174 > sofia/external/sipp at 185.35.228.51:5060 Standard REPORTING, cause: > NORMAL_CLEARING > 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:938 ( > sofia/external/sipp at 185.35.228.51:5060) State REPORTING going to sleep > 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:610 ( > sofia/external/sipp at 185.35.228.51:5060) State Change CS_REPORTING -> > CS_DESTROY > 2017-06-18 22:26:34.339569 [DEBUG] switch_core_session.c:1664 Session 67 ( > sofia/external/sipp at 185.35.228.51:5060) Locked, Waiting on external entiti > 2017-06-18 22:26:34.339569 [NOTICE] switch_core_session.c:1682 Session 67 ( > sofia/external/sipp at 185.35.228.51:5060) Ended > 2017-06-18 22:26:34.339569 [NOTICE] switch_core_session.c:1686 Close > Channel sofia/external/sipp at 185.35.228.51:5060 [CS_DESTROY] > 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:741 ( > sofia/external/sipp at 185.35.228.51:5060) Running State Change CS_DESTROY > (Cur 0 Tot 68) > 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:751 ( > sofia/external/sipp at 185.35.228.51:5060) State DESTROY > 2017-06-18 22:26:34.339569 [DEBUG] mod_sofia.c:343 sofia/ > external/sipp at 185.35.228.51:5060 SOFIA DESTROY > 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:181 > sofia/external/sipp at 185.35.228.51:5060 Standard DESTROY > 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:751 ( > sofia/external/sipp at 185.35.228.51:5060) State DESTROY going to sleep > > > U 185.35.228.51:5060 -> 185.35.228.40:5080 > INVITE sip:441554555666 at 185.35.228.40:5080 SIP/2.0. > Via: SIP/2.0/UDP 185.35.228.51:5060;branch=z9hG4bK-27036-1-0. > From: sipp ;tag=27036SIPpTag001. > To: 441554555666 . > Call-ID: 1-27036 at 185.35.228.51. > CSeq: 1 INVITE. > Contact: sip:sipp at 185.35.228.51:5060. > Max-Forwards: 70. > Subject: Performance Test. > Content-Type: application/sdp. > Content-Length: 137. > . > v=0. > o=user1 53655765 2353687637 IN IP4 185.35.228.51. > s=-. > c=IN IP4 185.35.228.51. > t=0 0. > m=audio 6000 RTP/AVP 0. > a=rtpmap:0 PCMU/8000. > > # > U 185.35.228.40:5080 -> 185.35.228.51:5060 > SIP/2.0 100 Trying. > Via: SIP/2.0/UDP 185.35.228.51:5060;branch=z9hG4bK-27036-1-0. > From: sipp ;tag=27036SIPpTag001. > To: 441554555666 . > Call-ID: 1-27036 at 185.35.228.51. > CSeq: 1 INVITE. > User-Agent: FreeSWITCH-mod_sofia/1.6.18+git~20170612T211449Z~ > 6e79667c0a~64bit. > Content-Length: 0. > . > > # > U 185.35.228.40:5080 -> 185.35.228.51:5060 > SIP/2.0 180 Ringing. > Via: SIP/2.0/UDP 185.35.228.51:5060;branch=z9hG4bK-27036-1-0. > From: sipp ;tag=27036SIPpTag001. > To: 441554555666 ;tag=91vp8601aS4Qp. > Call-ID: 1-27036 at 185.35.228.51. > CSeq: 1 INVITE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.6.18+git~20170612T211449Z~ > 6e79667c0a~64bit. > Accept: application/sdp. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, > REFER, NOTIFY. > Supported: timer, path, replaces. > Allow-Events: talk, hold, conference, refer. > Content-Length: 0. > Remote-Party-ID: "Outbound Call" >;party=calling;privacy=off;screen=no. > . > > # > U 185.35.228.40:5080 -> 185.35.228.51:5060 > SIP/2.0 200 OK. > Via: SIP/2.0/UDP 185.35.228.51:5060;branch=z9hG4bK-27036-1-0. > From: sipp ;tag=27036SIPpTag001. > To: 441554555666 ;tag=91vp8601aS4Qp. > Call-ID: 1-27036 at 185.35.228.51. > CSeq: 1 INVITE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.6.18+git~20170612T211449Z~ > 6e79667c0a~64bit. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, > REFER, NOTIFY. > Supported: timer, path, replaces. > Allow-Events: talk, hold, conference, refer. > Content-Type: application/sdp. > Content-Disposition: session. > Content-Length: 166. > Remote-Party-ID: "Outbound Call" >;party=calling;privacy=off;screen=no. > . > v=0. > o=FreeSWITCH 1497797499 1497797500 IN IP4 185.35.228.40. > s=FreeSWITCH. > c=IN IP4 185.35.228.40. > t=0 0. > m=audio 25252 RTP/AVP 0. > a=rtpmap:0 PCMU/8000. > a=ptime:20. > > # > U 185.35.228.51:5060 -> 185.35.228.40:5080 > ACK sip:441554555666 at 185.35.228.40:5080 SIP/2.0. > Via: SIP/2.0/UDP 185.35.228.51:5060;branch=z9hG4bK-27036-1-5. > From: sipp ;tag=27036SIPpTag001. > To: 441554555666 ;tag=91vp8601aS4Qp. > Call-ID: 1-27036 at 185.35.228.51. > CSeq: 1 ACK. > Contact: sip:sipp at 185.35.228.51:5060. > Max-Forwards: 70. > Subject: Performance Test. > Content-Length: 0. > . > > # > U 185.35.228.40:5080 -> 185.35.228.51:5060 > BYE sip:sipp at 185.35.228.51:5060 SIP/2.0. > Via: SIP/2.0/UDP 185.35.228.40:5080;rport;branch=z9hG4bKUBe0HvDjHF1XN. > Max-Forwards: 70. > From: 441554555666 >;tag=91vp8601aS4Qp. > To: sipp ;tag=27036SIPpTag001. > Call-ID: 1-27036 at 185.35.228.51. > CSeq: 108575311 BYE. > User-Agent: FreeSWITCH-mod_sofia/1.6.18+git~20170612T211449Z~ > 6e79667c0a~64bit. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, > REFER, NOTIFY. > Supported: timer, path, replaces. > Reason: Q.850;cause=16;text="NORMAL_CLEARING". > Content-Length: 0. > . > > # > U 185.35.228.51:5060 -> 185.35.228.40:5080 > SIP/2.0 200 OK. > Via: SIP/2.0/UDP 185.35.228.40:5080;rport;branch=z9hG4bKUBe0HvDjHF1XN. > From: 441554555666 >;tag=91vp8601aS4Qp. > To: sipp ;tag=27036SIPpTag001. > Call-ID: 1-27036 at 185.35.228.51. > CSeq: 108575311 BYE. > Contact: . > Content-Length: 0. > > > > NGREP of SIP messages from FS to terminator > > > U 185.35.228.40:5080 -> 185.35.228.48:5060 > INVITE sip:441554555666 at 185.35.228.48:5060 SIP/2.0. > Via: SIP/2.0/UDP 185.35.228.40:5080;rport;branch=z9hG4bKXX0HNjFSB1D3c. > Max-Forwards: 69. > From: "sipp" ;tag=eFUjHeNKv5KNg. > To: . > Call-ID: c755e7c8-cf13-1235-25a9-363165383663. > CSeq: 108575420 INVITE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.6.18+git~20170612T211449Z~ > 6e79667c0a~64bit. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, > REFER, NOTIFY. > Supported: timer, path, replaces. > Allow-Events: talk, hold, conference, refer. > Content-Type: application/sdp. > Content-Disposition: session. > Content-Length: 222. > X-FS-Support: update_display,send_info. > Remote-Party-ID: "sipp" ;party=calling;screen=yes; > privacy=off. > . > v=0. > o=FreeSWITCH 1497801741 1497801743 IN IP4 185.35.228.40. > s=FreeSWITCH. > c=IN IP4 185.35.228.40. > t=0 0. > m=audio 21228 RTP/AVP 0 101. > a=rtpmap:0 PCMU/8000. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=ptime:20. > > # > U 185.35.228.48:5060 -> 185.35.228.40:5080 > SIP/2.0 180 Ringing. > Via: SIP/2.0/UDP 185.35.228.40:5080;rport;branch=z9hG4bKXX0HNjFSB1D3c. > From: "sipp" ;tag=eFUjHeNKv5KNg. > To: ;tag=31480SIPpTag018. > Call-ID: c755e7c8-cf13-1235-25a9-363165383663. > CSeq: 108575420 INVITE. > Contact: . > Content-Length: 0. > . > > # > U 185.35.228.48:5060 -> 185.35.228.40:5080 > SIP/2.0 200 OK. > Via: SIP/2.0/UDP 185.35.228.40:5080;rport;branch=z9hG4bKXX0HNjFSB1D3c. > From: "sipp" ;tag=eFUjHeNKv5KNg. > To: ;tag=31480SIPpTag018. > Call-ID: c755e7c8-cf13-1235-25a9-363165383663. > CSeq: 108575420 INVITE. > Contact: . > Content-Type: application/sdp. > Content-Length: 137. > . > v=0. > o=user1 53655765 2353687637 IN IP4 185.35.228.48. > s=-. > c=IN IP4 185.35.228.48. > t=0 0. > m=audio 6000 RTP/AVP 0. > a=rtpmap:0 PCMU/8000. > > # > U 185.35.228.40:5080 -> 185.35.228.48:5060 > ACK sip:185.35.228.48:5060;transport=UDP SIP/2.0. > Via: SIP/2.0/UDP 185.35.228.40:5080;rport;branch=z9hG4bKy6SaQD0v893Nr. > Max-Forwards: 70. > From: "sipp" ;tag=eFUjHeNKv5KNg. > To: ;tag=31480SIPpTag018. > Call-ID: c755e7c8-cf13-1235-25a9-363165383663. > CSeq: 108575420 ACK. > Contact: . > Content-Length: 0. > . > > # > U 185.35.228.40:5080 -> 185.35.228.48:5060 > BYE sip:185.35.228.48:5060;transport=UDP SIP/2.0. > Via: SIP/2.0/UDP 185.35.228.40:5080;rport;branch=z9hG4bKZFK3r8g05jt8K. > Max-Forwards: 70. > From: "sipp" ;tag=eFUjHeNKv5KNg. > To: ;tag=31480SIPpTag018. > Call-ID: c755e7c8-cf13-1235-25a9-363165383663. > CSeq: 108575421 BYE. > User-Agent: FreeSWITCH-mod_sofia/1.6.18+git~20170612T211449Z~ > 6e79667c0a~64bit. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, > REFER, NOTIFY. > Supported: timer, path, replaces. > Reason: SIP;cause=487;text="ORIGINATOR_CANCEL". > Content-Length: 0. > . > > # > U 185.35.228.48:5060 -> 185.35.228.40:5080 > SIP/2.0 200 OK. > Via: SIP/2.0/UDP 185.35.228.40:5080;rport;branch=z9hG4bKZFK3r8g05jt8K. > From: "sipp" ;tag=eFUjHeNKv5KNg. > To: ;tag=31480SIPpTag018. > Call-ID: c755e7c8-cf13-1235-25a9-363165383663. > CSeq: 108575421 BYE. > Contact: . > Content-Length: 0. > . > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From ahmed at netelsat.net Sun Jun 18 23:33:23 2017 From: ahmed at netelsat.net (Ahmed Sboor) Date: Mon, 19 Jun 2017 04:33:23 +0500 Subject: [Freeswitch-users] Call Dropping In-Reply-To: References: <222F1F16-373C-484A-B290-200CB13F2CAD@tm.net.uk> Message-ID: if on VCS , Rate table is set and balance or credit limit is also positive and rate exist in rate table , then h323-credit-limit is also set. you should also post mod radius debug logs. On Mon, Jun 19, 2017 at 4:08 AM, Colin Morelli wrote: > You've got an execute_on_answer of sched_hangup(+${h323-credit-time} > alloted_timeout) > > Immediately after your call is answered: > 2017-06-18 22:26:34.279530 [NOTICE] switch_ivr_originate.c:3647 Channel [ > sofia/external/sipp at 185.35.228.51:5060] has been answered > EXECUTE sofia/external/sipp at 185.35.228.51:5060 sched_hangup(+ > alloted_timeout) > > It would seem that h323-credit-time is not being set, which is causing > sched_hangup to immediately hangup the call on answer. > > On Sun, Jun 18, 2017 at 6:52 PM, Joseph Waite wrote: > >> Hi Guys >> >> Using FreeSwitch with Radius linked to JeraSoft VCS billing system. >> >> I am sending a Call from a SIPP originator, through the FreeSwitch box >> and back out to another SIPP terminator scenario. >> The call goes through ok, everything happens as it should, however the >> call immediately drops, I have done egrep’s of both sides of the call and >> the BYE is defiantly coming from Freeswitch for some reason but I cannot >> work out why. Anyone any ideas? >> I am attaching the FreeSwitch logs plus the egrep’s >> If I register zipper on my laptop to FS and make a call works fine. >> >> 2017-06-18 22:26:34.199519 [NOTICE] switch_channel.c:1104 New Channel >> sofia/external/sipp at 185.35.228.51:5060 [15021010-8f64-439f-8dbb-1afe0 >> 90c44a5] >> 2017-06-18 22:26:34.199519 [DEBUG] switch_core_state_machine.c:584 ( >> sofia/external/sipp at 185.35.228.51:5060) Running State Change CS_NEW (Cur >> 1 Tot 67) >> 2017-06-18 22:26:34.199519 [DEBUG] sofia.c:9837 sofia/external/si >> pp at 185.35.228.51:5060 receiving invite from 185.35.228.51:5060 version: >> 1.6.18 git 6e79667 2017-06-12 21:14:49Z 64bit >> 2017-06-18 22:26:34.219492 [DEBUG] sofia.c:7048 Channel >> sofia/external/sipp at 185.35.228.51:5060 entering state [received][100] >> 2017-06-18 22:26:34.219492 [DEBUG] sofia.c:7058 Remote SDP: >> v=0 >> o=user1 53655765 2353687637 IN IP4 185.35.228.51 >> s=- >> c=IN IP4 185.35.228.51 >> t=0 0 >> m=audio 6000 RTP/AVP 0 >> a=rtpmap:0 PCMU/8000 >> >> 2017-06-18 22:26:34.219492 [DEBUG] sofia.c:7450 ( >> sofia/external/sipp at 185.35.228.51:5060) State Change CS_NEW -> CS_INIT >> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:603 ( >> sofia/external/sipp at 185.35.228.51:5060) State NEW >> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:584 ( >> sofia/external/sipp at 185.35.228.51:5060) Running State Change CS_INIT >> (Cur 1 Tot 67) >> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:627 ( >> sofia/external/sipp at 185.35.228.51:5060) State INIT >> 2017-06-18 22:26:34.219492 [DEBUG] mod_sofia.c:90 sofia/external/ >> sipp at 185.35.228.51:5060 SOFIA INIT >> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:40 >> sofia/external/sipp at 185.35.228.51:5060 Standard INIT >> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:48 ( >> sofia/external/sipp at 185.35.228.51:5060) State Change CS_INIT -> >> CS_ROUTING >> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:627 ( >> sofia/external/sipp at 185.35.228.51:5060) State INIT going to sleep >> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:584 ( >> sofia/external/sipp at 185.35.228.51:5060) Running State Change CS_ROUTING >> (Cur 1 Tot 67) >> 2017-06-18 22:26:34.219492 [DEBUG] switch_channel.c:2249 ( >> sofia/external/sipp at 185.35.228.51:5060) Callstate Change DOWN -> RINGING >> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:643 ( >> sofia/external/sipp at 185.35.228.51:5060) State ROUTING >> 2017-06-18 22:26:34.219492 [DEBUG] mod_sofia.c:143 sofia/external >> /sipp at 185.35.228.51:5060 SOFIA ROUTING >> 2017-06-18 22:26:34.219492 [ERR] mod_xml_radius.c:933 Result of true >> match: 185.35.228.40 == ^185\.35\.229\.30 >> 2017-06-18 22:26:34.219492 [INFO] mod_xml_radius.c:986 mod_xml_radius: >> Accounting Start success >> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:236 >> sofia/external/sipp at 185.35.228.51:5060 Standard ROUTING >> 2017-06-18 22:26:34.219492 [INFO] mod_dialplan_xml.c:637 Processing sipp >> ->441554555666 in context public >> Dialplan: sofia/external/sipp at 185.35.228.51:5060 parsing >> [public->unloop] continue=false >> Dialplan: sofia/external/sipp at 185.35.228.51:5060 Regex (PASS) [unloop] >> ${unroll_loops}(true) =~ /^true$/ break=on-false >> Dialplan: sofia/external/sipp at 185.35.228.51:5060 Regex (FAIL) [unloop] >> ${sip_looped_call}() =~ /^true$/ break=on-false >> Dialplan: sofia/external/sipp at 185.35.228.51:5060 parsing >> [public->outside_call] continue=true >> Dialplan: sofia/external/sipp at 185.35.228.51:5060 Absolute Condition >> [outside_call] >> Dialplan: sofia/external/sipp at 185.35.228.51:5060 Action >> set(outside_call=true) >> Dialplan: sofia/external/sipp at 185.35.228.51:5060 Action >> export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) >> Dialplan: sofia/external/sipp at 185.35.228.51:5060 parsing >> [public->call_debug] continue=true >> Dialplan: sofia/external/sipp at 185.35.228.51:5060 Regex (FAIL) >> [call_debug] ${call_debug}(false) =~ /^true$/ break=never >> Dialplan: sofia/external/sipp at 185.35.228.51:5060 parsing >> [public->rejections] continue=false >> Dialplan: sofia/external/sipp at 185.35.228.51:5060 Regex (FAIL) >> [rejections] ${radius_auth_result}() =~ /2/ break=on-false >> Dialplan: sofia/external/sipp at 185.35.228.51:5060 parsing >> [public->timedouts] continue=false >> Dialplan: sofia/external/sipp at 185.35.228.51:5060 Regex (FAIL) >> [timedouts] ${radius_auth_result}() =~ /1/ break=on-false >> Dialplan: sofia/external/sipp at 185.35.228.51:5060 parsing >> [public->JeraSoft VCS Routing] continue=false >> Dialplan: sofia/external/sipp at 185.35.228.51:5060 Regex (PASS) [JeraSoft >> VCS Routing] destination_number(441554555666) =~ /^(.+)$/ break=on-false >> Dialplan: sofia/external/sipp at 185.35.228.51:5060 Action >> export(nolocal:h323-call-origin=originate) >> Dialplan: sofia/external/sipp at 185.35.228.51:5060 Action >> set(sip_h_X-accountcode=${accountcode}) >> Dialplan: sofia/external/sipp at 185.35.228.51:5060 Action >> set(call_direction=outbound) >> Dialplan: sofia/external/sipp at 185.35.228.51:5060 Action >> set(hangup_after_bridge=true) >> Dialplan: sofia/external/sipp at 185.35.228.51:5060 Action >> set(continue_on_fail=true) >> Dialplan: sofia/external/sipp at 185.35.228.51:5060 Action >> set(inherit_codec=true) >> Dialplan: sofia/external/sipp at 185.35.228.51:5060 Action >> set(call_timeout=20) >> Dialplan: sofia/external/sipp at 185.35.228.51:5060 Action >> set(fail_on_single_reject=USER_BUSY) >> Dialplan: sofia/external/sipp at 185.35.228.51:5060 Action >> set(origination_caller_id_name=${sip_req_user}) >> Dialplan: sofia/external/sipp at 185.35.228.51:5060 Action >> set(origination_caller_id_number=${sip_from_user}) >> Dialplan: sofia/external/sipp at 185.35.228.51:5060 Action >> set(execute_on_answer=sched_hangup +${h323-credit-time} alloted_timeout) >> Dialplan: sofia/external/sipp at 185.35.228.51:5060 Action >> bridge({sip_invite_from_uri=sip:${sip_from_user}@${sip_netwo >> rk_ip}}sofia/external/${destination_number}@185.35.229.30:5060 >> ) >> Dialplan: sofia/external/sipp at 185.35.228.51:5060 Action >> hangup(${bridge_hangup_cause}) >> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:286 ( >> sofia/external/sipp at 185.35.228.51:5060) State Change CS_ROUTING -> >> CS_EXECUTE >> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:643 ( >> sofia/external/sipp at 185.35.228.51:5060) State ROUTING going to sleep >> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:584 ( >> sofia/external/sipp at 185.35.228.51:5060) Running State Change CS_EXECUTE >> (Cur 1 Tot 67) >> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:650 ( >> sofia/external/sipp at 185.35.228.51:5060) State EXECUTE >> 2017-06-18 22:26:34.219492 [DEBUG] mod_sofia.c:198 sofia/external >> /sipp at 185.35.228.51:5060 SOFIA EXECUTE >> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:328 >> sofia/external/sipp at 185.35.228.51:5060 Standard EXECUTE >> EXECUTE sofia/external/sipp at 185.35.228.51:5060 set(outside_call=true) >> 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET >> sofia/external/sipp at 185.35.228.51:5060 [outside_call]=[true] >> EXECUTE sofia/external/sipp at 185.35.228.51:5060 export(RFC2822_DATE=Sun, >> 18 Jun 2017 22:26:34 +0100) >> 2017-06-18 22:26:34.219492 [DEBUG] switch_channel.c:1296 EXPORT >> (export_vars) [RFC2822_DATE]=[Sun, 18 Jun 2017 22:26:34 +0100] >> EXECUTE sofia/external/sipp at 185.35.228.51:5060 >> export(nolocal:h323-call-origin=originate) >> 2017-06-18 22:26:34.219492 [DEBUG] switch_channel.c:1296 EXPORT >> (export_vars) (REMOTE ONLY) [h323-call-origin]=[originate] >> EXECUTE sofia/external/sipp at 185.35.228.51:5060 set(sip_h_X-accountcode=) >> 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET >> sofia/external/sipp at 185.35.228.51:5060 [sip_h_X-accountcode]=[UNDEF] >> EXECUTE sofia/external/sipp at 185.35.228.51:5060 >> set(call_direction=outbound) >> 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET >> sofia/external/sipp at 185.35.228.51:5060 [call_direction]=[outbound] >> EXECUTE sofia/external/sipp at 185.35.228.51:5060 >> set(hangup_after_bridge=true) >> 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET >> sofia/external/sipp at 185.35.228.51:5060 [hangup_after_bridge]=[true] >> EXECUTE sofia/external/sipp at 185.35.228.51:5060 set(continue_on_fail=true) >> 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET >> sofia/external/sipp at 185.35.228.51:5060 [continue_on_fail]=[true] >> EXECUTE sofia/external/sipp at 185.35.228.51:5060 set(inherit_codec=true) >> 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET >> sofia/external/sipp at 185.35.228.51:5060 [inherit_codec]=[true] >> EXECUTE sofia/external/sipp at 185.35.228.51:5060 set(call_timeout=20) >> 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET >> sofia/external/sipp at 185.35.228.51:5060 [call_timeout]=[20] >> EXECUTE sofia/external/sipp at 185.35.228.51:5060 >> set(fail_on_single_reject=USER_BUSY) >> 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET >> sofia/external/sipp at 185.35.228.51:5060 [fail_on_single_reject]=[USER_ >> BUSY] >> EXECUTE sofia/external/sipp at 185.35.228.51:5060 >> set(origination_caller_id_name=441554555666) >> 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET >> sofia/external/sipp at 185.35.228.51:5060 [origination_caller_id_name]=[ >> 441554555666] >> EXECUTE sofia/external/sipp at 185.35.228.51:5060 >> set(origination_caller_id_number=sipp) >> 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET >> sofia/external/sipp at 185.35.228.51:5060 [origination_caller_id_number] >> =[sipp] >> EXECUTE sofia/external/sipp at 185.35.228.51:5060 >> set(execute_on_answer=sched_hangup + alloted_timeout) >> 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET >> sofia/external/sipp at 185.35.228.51:5060 [execute_on_answer]=[sched_hangup >> + alloted_timeout] >> EXECUTE sofia/external/sipp at 185.35.228.51:5060 >> bridge({sip_invite_from_uri=sip:sipp at 185.35.228.51}sofia/ext >> ernal/441554555666 at 185.35.229.30:5060) >> 2017-06-18 22:26:34.219492 [DEBUG] switch_channel.c:1250 sofia/ex >> ternal/sipp at 185.35.228.51:5060 EXPORTING[export_vars] >> [RFC2822_DATE]=[Sun, 18 Jun 2017 22:26:34 +0100] to event >> 2017-06-18 22:26:34.219492 [DEBUG] switch_channel.c:1250 sofia/ex >> ternal/sipp at 185.35.228.51:5060 EXPORTING[export_vars] >> [h323-call-origin]=[originate] to event >> 2017-06-18 22:26:34.219492 [DEBUG] switch_ivr_originate.c:2142 Parsing >> global variables >> 2017-06-18 22:26:34.219492 [NOTICE] switch_channel.c:1104 New Channel >> sofia/external/441554555666 at 185.35.229.30:5060 >> [96c1a021-5195-41ce-b903-08b98816d70d] >> 2017-06-18 22:26:34.219492 [DEBUG] mod_sofia.c:4819 ( >> sofia/external/441554555666 at 185.35.229.30:5060) State Change CS_NEW -> >> CS_INIT >> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:584 ( >> sofia/external/441554555666 at 185.35.229.30:5060) Running State Change >> CS_INIT (Cur 2 Tot 68) >> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:627 ( >> sofia/external/441554555666 at 185.35.229.30:5060) State INIT >> 2017-06-18 22:26:34.219492 [DEBUG] mod_sofia.c:90 sofia/external/ >> 441554555666 at 185.35.229.30:5060 SOFIA INIT >> 2017-06-18 22:26:34.219492 [DEBUG] sofia_glue.c:1295 sofia/extern >> al/441554555666 at 185.35.229.30:5060 sending invite version: 1.6.18 git >> 6e79667 2017-06-12 21:14:49Z 64bit >> Local SDP: >> v=0 >> o=FreeSWITCH 1497789362 1497789363 IN IP4 185.35.228.40 >> s=FreeSWITCH >> c=IN IP4 185.35.228.40 >> t=0 0 >> m=audio 31832 RTP/AVP 0 101 13 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=rtpmap:13 CN/8000 >> a=ptime:20 >> a=sendrecv >> >> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:40 >> sofia/external/441554555666 at 185.35.229.30:5060 Standard INIT >> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:48 ( >> sofia/external/441554555666 at 185.35.229.30:5060) State Change CS_INIT -> >> CS_ROUTING >> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:627 ( >> sofia/external/441554555666 at 185.35.229.30:5060) State INIT going to sleep >> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:584 ( >> sofia/external/441554555666 at 185.35.229.30:5060) Running State Change >> CS_ROUTING (Cur 2 Tot 68) >> 2017-06-18 22:26:34.219492 [DEBUG] sofia.c:7048 Channel >> sofia/external/441554555666 at 185.35.229.30:5060 entering state >> [calling][0] >> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:643 ( >> sofia/external/441554555666 at 185.35.229.30:5060) State ROUTING >> 2017-06-18 22:26:34.219492 [DEBUG] mod_sofia.c:143 sofia/external >> /441554555666 at 185.35.229.30:5060 SOFIA ROUTING >> 2017-06-18 22:26:34.219492 [DEBUG] switch_ivr_originate.c:67 ( >> sofia/external/441554555666 at 185.35.229.30:5060) State Change CS_ROUTING >> -> CS_CONSUME_MEDIA >> 2017-06-18 22:26:34.219492 [ERR] mod_xml_radius.c:930 Didn't match: >> 185.35.229.30:5060 == ^185\.35\.229\.30 >> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:643 ( >> sofia/external/441554555666 at 185.35.229.30:5060) State ROUTING going to >> sleep >> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:584 ( >> sofia/external/441554555666 at 185.35.229.30:5060) Running State Change >> CS_CONSUME_MEDIA (Cur 2 Tot 68) >> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:662 ( >> sofia/external/441554555666 at 185.35.229.30:5060) State CONSUME_MEDIA >> 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:662 ( >> sofia/external/441554555666 at 185.35.229.30:5060) State CONSUME_MEDIA >> going to sleep >> 2017-06-18 22:26:34.259480 [DEBUG] sofia_glue.c:1295 sofia/extern >> al/441554555666 at 185.35.229.30:5060 sending invite version: 1.6.18 git >> 6e79667 2017-06-12 21:14:49Z 64bit >> Local SDP: >> v=0 >> o=FreeSWITCH 1497789362 1497789364 IN IP4 185.35.228.40 >> s=FreeSWITCH >> c=IN IP4 185.35.228.40 >> t=0 0 >> m=audio 31832 RTP/AVP 0 101 >> a=rtpmap:0 PCMU/8000 >> a=rtpmap:101 telephone-event/8000 >> a=fmtp:101 0-16 >> a=ptime:20 >> a=sendrecv >> >> 2017-06-18 22:26:34.259480 [DEBUG] sofia.c:7048 Channel >> sofia/external/441554555666 at 185.35.229.30:5060 entering state >> [calling][0] >> 2017-06-18 22:26:34.259480 [DEBUG] sofia.c:7048 Channel >> sofia/external/441554555666 at 185.35.229.30:5060 entering state >> [proceeding][180] >> 2017-06-18 22:26:34.259480 [NOTICE] sofia.c:7156 Ring-Ready >> sofia/external/441554555666 at 185.35.229.30:5060! >> 2017-06-18 22:26:34.259480 [DEBUG] switch_channel.c:3346 ( >> sofia/external/441554555666 at 185.35.229.30:5060) Callstate Change DOWN -> >> RINGING >> 2017-06-18 22:26:34.259480 [DEBUG] sofia.c:7048 Channel >> sofia/external/441554555666 at 185.35.229.30:5060 entering state >> [completing][200] >> 2017-06-18 22:26:34.259480 [DEBUG] sofia.c:7058 Remote SDP: >> v=0 >> o=user1 53655765 2353687637 IN IP4 185.35.228.48 >> s=- >> c=IN IP4 185.35.228.48 >> t=0 0 >> m=audio 6000 RTP/AVP 0 >> a=rtpmap:0 PCMU/8000 >> >> 2017-06-18 22:26:34.259480 [DEBUG] sofia.c:7048 Channel >> sofia/external/441554555666 at 185.35.229.30:5060 entering state >> [ready][200] >> 2017-06-18 22:26:34.259480 [DEBUG] switch_core_media.c:4445 Audio Codec >> Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] >> 2017-06-18 22:26:34.259480 [DEBUG] switch_core_media.c:4500 Audio Codec >> Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match >> 2017-06-18 22:26:34.259480 [DEBUG] switch_core_media.c:3057 Set Codec >> sofia/external/441554555666 at 185.35.229.30:5060 PCMU/8000 20 ms 160 >> samples 64000 bits 1 channels >> 2017-06-18 22:26:34.259480 [DEBUG] switch_core_codec.c:111 sofia/ >> external/441554555666 at 185.35.229.30:5060 Original read codec set to >> PCMU:0 >> 2017-06-18 22:26:34.259480 [DEBUG] switch_core_media.c:4770 No 2833 in >> SDP. Disable 2833 dtmf and switch to INFO >> 2017-06-18 22:26:34.259480 [DEBUG] switch_core_media.c:6874 AUDIO RTP [ >> sofia/external/441554555666 at 185.35.229.30:5060] 185.35.228.40 port 31832 >> -> 185.35.228.48 port 6000 codec: 0 ms: 20 >> 2017-06-18 22:26:34.259480 [DEBUG] switch_rtp.c:4108 Starting timer >> [soft] 160 bytes per 20ms >> 2017-06-18 22:26:34.279530 [DEBUG] switch_core_media.c:7205 sofia >> /external/441554555666 at 185.35.229.30:5060 Set rtp dtmf delay to 40 >> 2017-06-18 22:26:34.279530 [NOTICE] sofia.c:8182 Channel [ >> sofia/external/441554555666 at 185.35.229.30:5060] has been answered >> 2017-06-18 22:26:34.279530 [DEBUG] switch_channel.c:3773 ( >> sofia/external/441554555666 at 185.35.229.30:5060) Callstate Change RINGING >> -> ACTIVE >> 2017-06-18 22:26:34.279530 [NOTICE] mod_sofia.c:2273 Ring-Ready >> sofia/external/sipp at 185.35.228.51:5060! >> 2017-06-18 22:26:34.279530 [DEBUG] sofia.c:7048 Channel >> sofia/external/sipp at 185.35.228.51:5060 entering state [early][180] >> 2017-06-18 22:26:34.279530 [NOTICE] switch_ivr_originate.c:525 Ring Ready >> sofia/external/sipp at 185.35.228.51:5060! >> 2017-06-18 22:26:34.279530 [DEBUG] switch_ivr_originate.c:410 Setting >> codec string on sofia/external/sipp at 185.35.228.51:5060 to PCMU at 8000h@20i >> 2017-06-18 22:26:34.279530 [DEBUG] switch_core_media.c:4445 Audio Codec >> Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] >> 2017-06-18 22:26:34.279530 [DEBUG] switch_core_media.c:4500 Audio Codec >> Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match >> 2017-06-18 22:26:34.279530 [DEBUG] switch_core_media.c:3057 Set Codec >> sofia/external/sipp at 185.35.228.51:5060 PCMU/8000 20 ms 160 samples 64000 >> bits 1 channels >> 2017-06-18 22:26:34.279530 [DEBUG] switch_core_codec.c:111 sofia/ >> external/sipp at 185.35.228.51:5060 Original read codec set to PCMU:0 >> 2017-06-18 22:26:34.279530 [DEBUG] switch_core_media.c:4770 No 2833 in >> SDP. Disable 2833 dtmf and switch to INFO >> 2017-06-18 22:26:34.279530 [DEBUG] switch_core_media.c:6874 AUDIO RTP [ >> sofia/external/sipp at 185.35.228.51:5060] 185.35.228.40 port 23728 -> >> 185.35.228.51 port 6000 codec: 0 ms: 20 >> 2017-06-18 22:26:34.279530 [DEBUG] switch_rtp.c:4108 Starting timer >> [soft] 160 bytes per 20ms >> 2017-06-18 22:26:34.279530 [DEBUG] switch_core_media.c:7205 sofia >> /external/sipp at 185.35.228.51:5060 Set rtp dtmf delay to 40 >> 2017-06-18 22:26:34.279530 [NOTICE] sofia_media.c:92 Pre-Answer >> sofia/external/sipp at 185.35.228.51:5060! >> 2017-06-18 22:26:34.279530 [DEBUG] switch_channel.c:3474 ( >> sofia/external/sipp at 185.35.228.51:5060) Callstate Change RINGING -> EARLY >> 2017-06-18 22:26:34.279530 [DEBUG] switch_core_media.c:6857 Audio params >> are unchanged for sofia/external/sipp at 185.35.228.51:5060. >> 2017-06-18 22:26:34.279530 [DEBUG] mod_sofia.c:850 Local SDP >> sofia/external/sipp at 185.35.228.51:5060: >> v=0 >> o=FreeSWITCH 1497797466 1497797467 IN IP4 185.35.228.40 >> s=FreeSWITCH >> c=IN IP4 185.35.228.40 >> t=0 0 >> m=audio 23728 RTP/AVP 0 >> a=rtpmap:0 PCMU/8000 >> a=ptime:20 >> a=sendrecv >> >> 2017-06-18 22:26:34.279530 [NOTICE] switch_ivr_originate.c:3647 Channel [ >> sofia/external/sipp at 185.35.228.51:5060] has been answered >> EXECUTE sofia/external/sipp at 185.35.228.51:5060 sched_hangup(+ >> alloted_timeout) >> 2017-06-18 22:26:34.279530 [DEBUG] sofia.c:7048 Channel >> sofia/external/sipp at 185.35.228.51:5060 entering state [completed][200] >> 2017-06-18 22:26:34.279530 [DEBUG] sofia.c:7048 Channel >> sofia/external/sipp at 185.35.228.51:5060 entering state [ready][200] >> 2017-06-18 22:26:34.279530 [NOTICE] mod_dptools.c:1188 Hangup >> sofia/external/sipp at 185.35.228.51:5060 [CS_EXECUTE] [NORMAL_CLEARING] >> 2017-06-18 22:26:34.279530 [DEBUG] switch_core_session.c:2814 sof >> ia/external/sipp at 185.35.228.51:5060 skip receive message >> [APPLICATION_EXEC_COMPLETE] (channel is hungup already) >> 2017-06-18 22:26:34.279530 [DEBUG] switch_channel.c:3773 ( >> sofia/external/sipp at 185.35.228.51:5060) Callstate Change EARLY -> ACTIVE >> 2017-06-18 22:26:34.279530 [DEBUG] switch_ivr_originate.c:3647 so >> fia/external/sipp at 185.35.228.51:5060 skip receive message [ANSWER_EVENT] >> (channel is hungup already) >> 2017-06-18 22:26:34.319525 [DEBUG] switch_ivr_originate.c:3848 Originate >> Resulted in Error Cause: 487 [ORIGINATOR_CANCEL] >> 2017-06-18 22:26:34.319525 [NOTICE] switch_ivr_originate.c:3938 Hangup >> sofia/external/441554555666 at 185.35.229.30:5060 [CS_CONSUME_MEDIA] >> [ORIGINATOR_CANCEL] >> 2017-06-18 22:26:34.319525 [DEBUG] switch_core_state_machine.c:584 ( >> sofia/external/441554555666 at 185.35.229.30:5060) Running State Change >> CS_HANGUP (Cur 2 Tot 68) >> 2017-06-18 22:26:34.319525 [DEBUG] switch_core_state_machine.c:850 ( >> sofia/external/441554555666 at 185.35.229.30:5060) Callstate Change ACTIVE >> -> HANGUP >> 2017-06-18 22:26:34.319525 [DEBUG] switch_core_state_machine.c:852 ( >> sofia/external/441554555666 at 185.35.229.30:5060) State HANGUP >> 2017-06-18 22:26:34.319525 [DEBUG] mod_sofia.c:438 Channel >> sofia/external/441554555666 at 185.35.229.30:5060 hanging up, cause: >> ORIGINATOR_CANCEL >> 2017-06-18 22:26:34.319525 [INFO] mod_dptools.c:3418 Originate Failed. >> Cause: ORIGINATOR_CANCEL >> 2017-06-18 22:26:34.319525 [DEBUG] mod_sofia.c:491 Sending BYE to >> sofia/external/441554555666 at 185.35.229.30:5060 >> 2017-06-18 22:26:34.319525 [DEBUG] switch_core_state_machine.c:60 >> sofia/external/441554555666 at 185.35.229.30:5060 Standard HANGUP, cause: >> ORIGINATOR_CANCEL >> 2017-06-18 22:26:34.319525 [DEBUG] switch_core_state_machine.c:852 ( >> sofia/external/441554555666 at 185.35.229.30:5060) State HANGUP going to >> sleep >> 2017-06-18 22:26:34.319525 [DEBUG] switch_core_state_machine.c:619 ( >> sofia/external/441554555666 at 185.35.229.30:5060) State Change CS_HANGUP >> -> CS_REPORTING >> 2017-06-18 22:26:34.319525 [DEBUG] switch_core_session.c:2814 sof >> ia/external/sipp at 185.35.228.51:5060 skip receive message >> [APPLICATION_EXEC_COMPLETE] (channel is hungup already) >> 2017-06-18 22:26:34.319525 [DEBUG] switch_core_state_machine.c:584 ( >> sofia/external/441554555666 at 185.35.229.30:5060) Running State Change >> CS_REPORTING (Cur 2 Tot 68) >> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:650 ( >> sofia/external/sipp at 185.35.228.51:5060) State EXECUTE going to sleep >> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:584 ( >> sofia/external/sipp at 185.35.228.51:5060) Running State Change CS_HANGUP >> (Cur 2 Tot 68) >> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:938 ( >> sofia/external/441554555666 at 185.35.229.30:5060) State REPORTING >> 2017-06-18 22:26:34.339569 [ERR] mod_xml_radius.c:930 Didn't match: >> 185.35.229.30 == ^185\.35\.229\.30 >> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:850 ( >> sofia/external/sipp at 185.35.228.51:5060) Callstate Change ACTIVE -> HANGUP >> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:174 >> sofia/external/441554555666 at 185.35.229.30:5060 Standard REPORTING, >> cause: ORIGINATOR_CANCEL >> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:938 ( >> sofia/external/441554555666 at 185.35.229.30:5060) State REPORTING going to >> sleep >> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:852 ( >> sofia/external/sipp at 185.35.228.51:5060) State HANGUP >> 2017-06-18 22:26:34.339569 [DEBUG] mod_sofia.c:438 Channel >> sofia/external/sipp at 185.35.228.51:5060 hanging up, cause: NORMAL_CLEARING >> 2017-06-18 22:26:34.339569 [DEBUG] mod_sofia.c:491 Sending BYE to >> sofia/external/sipp at 185.35.228.51:5060 >> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:60 >> sofia/external/sipp at 185.35.228.51:5060 Standard HANGUP, cause: >> NORMAL_CLEARING >> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:852 ( >> sofia/external/sipp at 185.35.228.51:5060) State HANGUP going to sleep >> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:619 ( >> sofia/external/sipp at 185.35.228.51:5060) State Change CS_HANGUP -> >> CS_REPORTING >> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:584 ( >> sofia/external/sipp at 185.35.228.51:5060) Running State Change >> CS_REPORTING (Cur 2 Tot 68) >> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:938 ( >> sofia/external/sipp at 185.35.228.51:5060) State REPORTING >> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:610 ( >> sofia/external/441554555666 at 185.35.229.30:5060) State Change >> CS_REPORTING -> CS_DESTROY >> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_session.c:1664 Session 68 ( >> sofia/external/441554555666 at 185.35.229.30:5060) Locked, Waiting on >> external entities >> 2017-06-18 22:26:34.339569 [ERR] mod_xml_radius.c:933 Result of true >> match: 185.35.228.40 == ^185\.35\.229\.30 >> 2017-06-18 22:26:34.339569 [NOTICE] switch_core_session.c:1682 Session 68 >> (sofia/external/441554555666 at 185.35.229.30:5060) Ended >> 2017-06-18 22:26:34.339569 [NOTICE] switch_core_session.c:1686 Close >> Channel sofia/external/441554555666 at 185.35.229.30:5060 [CS_DESTROY] >> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:741 ( >> sofia/external/441554555666 at 185.35.229.30:5060) Running State Change >> CS_DESTROY (Cur 1 Tot 68) >> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:751 ( >> sofia/external/441554555666 at 185.35.229.30:5060) State DESTROY >> 2017-06-18 22:26:34.339569 [DEBUG] mod_sofia.c:343 sofia/external >> /441554555666 at 185.35.229.30:5060 SOFIA DESTROY >> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:181 >> sofia/external/441554555666 at 185.35.229.30:5060 Standard DESTROY >> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:751 ( >> sofia/external/441554555666 at 185.35.229.30:5060) State DESTROY going to >> sleep >> 2017-06-18 22:26:34.339569 [INFO] mod_xml_radius.c:1044 mod_xml_radius: >> Accounting Stop success >> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:174 >> sofia/external/sipp at 185.35.228.51:5060 Standard REPORTING, cause: >> NORMAL_CLEARING >> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:938 ( >> sofia/external/sipp at 185.35.228.51:5060) State REPORTING going to sleep >> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:610 ( >> sofia/external/sipp at 185.35.228.51:5060) State Change CS_REPORTING -> >> CS_DESTROY >> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_session.c:1664 Session 67 ( >> sofia/external/sipp at 185.35.228.51:5060) Locked, Waiting on external >> entiti >> 2017-06-18 22:26:34.339569 [NOTICE] switch_core_session.c:1682 Session 67 >> (sofia/external/sipp at 185.35.228.51:5060) Ended >> 2017-06-18 22:26:34.339569 [NOTICE] switch_core_session.c:1686 Close >> Channel sofia/external/sipp at 185.35.228.51:5060 [CS_DESTROY] >> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:741 ( >> sofia/external/sipp at 185.35.228.51:5060) Running State Change CS_DESTROY >> (Cur 0 Tot 68) >> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:751 ( >> sofia/external/sipp at 185.35.228.51:5060) State DESTROY >> 2017-06-18 22:26:34.339569 [DEBUG] mod_sofia.c:343 sofia/external >> /sipp at 185.35.228.51:5060 SOFIA DESTROY >> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:181 >> sofia/external/sipp at 185.35.228.51:5060 Standard DESTROY >> 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:751 ( >> sofia/external/sipp at 185.35.228.51:5060) State DESTROY going to sleep >> >> >> U 185.35.228.51:5060 -> 185.35.228.40:5080 >> INVITE sip:441554555666 at 185.35.228.40:5080 SIP/2.0. >> Via: SIP/2.0/UDP 185.35.228.51:5060;branch=z9hG4bK-27036-1-0. >> From: sipp ;tag=27036SIPpTag001. >> To: 441554555666 . >> Call-ID: 1-27036 at 185.35.228.51. >> CSeq: 1 INVITE. >> Contact: sip:sipp at 185.35.228.51:5060. >> Max-Forwards: 70. >> Subject: Performance Test. >> Content-Type: application/sdp. >> Content-Length: 137. >> . >> v=0. >> o=user1 53655765 2353687637 IN IP4 185.35.228.51. >> s=-. >> c=IN IP4 185.35.228.51. >> t=0 0. >> m=audio 6000 RTP/AVP 0. >> a=rtpmap:0 PCMU/8000. >> >> # >> U 185.35.228.40:5080 -> 185.35.228.51:5060 >> SIP/2.0 100 Trying. >> Via: SIP/2.0/UDP 185.35.228.51:5060;branch=z9hG4bK-27036-1-0. >> From: sipp ;tag=27036SIPpTag001. >> To: 441554555666 . >> Call-ID: 1-27036 at 185.35.228.51. >> CSeq: 1 INVITE. >> User-Agent: FreeSWITCH-mod_sofia/1.6.18+git~20170612T211449Z~6e79667c0a~ >> 64bit. >> Content-Length: 0. >> . >> >> # >> U 185.35.228.40:5080 -> 185.35.228.51:5060 >> SIP/2.0 180 Ringing. >> Via: SIP/2.0/UDP 185.35.228.51:5060;branch=z9hG4bK-27036-1-0. >> From: sipp ;tag=27036SIPpTag001. >> To: 441554555666 ;tag=91vp8601aS4Qp. >> Call-ID: 1-27036 at 185.35.228.51. >> CSeq: 1 INVITE. >> Contact: . >> User-Agent: FreeSWITCH-mod_sofia/1.6.18+git~20170612T211449Z~6e79667c0a~ >> 64bit. >> Accept: application/sdp. >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, >> REGISTER, REFER, NOTIFY. >> Supported: timer, path, replaces. >> Allow-Events: talk, hold, conference, refer. >> Content-Length: 0. >> Remote-Party-ID: "Outbound Call" > >;party=calling;privacy=off;screen=no. >> . >> >> # >> U 185.35.228.40:5080 -> 185.35.228.51:5060 >> SIP/2.0 200 OK. >> Via: SIP/2.0/UDP 185.35.228.51:5060;branch=z9hG4bK-27036-1-0. >> From: sipp ;tag=27036SIPpTag001. >> To: 441554555666 ;tag=91vp8601aS4Qp. >> Call-ID: 1-27036 at 185.35.228.51. >> CSeq: 1 INVITE. >> Contact: . >> User-Agent: FreeSWITCH-mod_sofia/1.6.18+git~20170612T211449Z~6e79667c0a~ >> 64bit. >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, >> REGISTER, REFER, NOTIFY. >> Supported: timer, path, replaces. >> Allow-Events: talk, hold, conference, refer. >> Content-Type: application/sdp. >> Content-Disposition: session. >> Content-Length: 166. >> Remote-Party-ID: "Outbound Call" > >;party=calling;privacy=off;screen=no. >> . >> v=0. >> o=FreeSWITCH 1497797499 1497797500 IN IP4 185.35.228.40. >> s=FreeSWITCH. >> c=IN IP4 185.35.228.40. >> t=0 0. >> m=audio 25252 RTP/AVP 0. >> a=rtpmap:0 PCMU/8000. >> a=ptime:20. >> >> # >> U 185.35.228.51:5060 -> 185.35.228.40:5080 >> ACK sip:441554555666 at 185.35.228.40:5080 SIP/2.0. >> Via: SIP/2.0/UDP 185.35.228.51:5060;branch=z9hG4bK-27036-1-5. >> From: sipp ;tag=27036SIPpTag001. >> To: 441554555666 ;tag=91vp8601aS4Qp. >> Call-ID: 1-27036 at 185.35.228.51. >> CSeq: 1 ACK. >> Contact: sip:sipp at 185.35.228.51:5060. >> Max-Forwards: 70. >> Subject: Performance Test. >> Content-Length: 0. >> . >> >> # >> U 185.35.228.40:5080 -> 185.35.228.51:5060 >> BYE sip:sipp at 185.35.228.51:5060 SIP/2.0. >> Via: SIP/2.0/UDP 185.35.228.40:5080;rport;branch=z9hG4bKUBe0HvDjHF1XN. >> Max-Forwards: 70. >> From: 441554555666 > >;tag=91vp8601aS4Qp. >> To: sipp ;tag=27036SIPpTag001. >> Call-ID: 1-27036 at 185.35.228.51. >> CSeq: 108575311 BYE. >> User-Agent: FreeSWITCH-mod_sofia/1.6.18+git~20170612T211449Z~6e79667c0a~ >> 64bit. >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, >> REGISTER, REFER, NOTIFY. >> Supported: timer, path, replaces. >> Reason: Q.850;cause=16;text="NORMAL_CLEARING". >> Content-Length: 0. >> . >> >> # >> U 185.35.228.51:5060 -> 185.35.228.40:5080 >> SIP/2.0 200 OK. >> Via: SIP/2.0/UDP 185.35.228.40:5080;rport;branch=z9hG4bKUBe0HvDjHF1XN. >> From: 441554555666 > >;tag=91vp8601aS4Qp. >> To: sipp ;tag=27036SIPpTag001. >> Call-ID: 1-27036 at 185.35.228.51. >> CSeq: 108575311 BYE. >> Contact: . >> Content-Length: 0. >> >> >> >> NGREP of SIP messages from FS to terminator >> >> >> U 185.35.228.40:5080 -> 185.35.228.48:5060 >> INVITE sip:441554555666 at 185.35.228.48:5060 SIP/2.0. >> Via: SIP/2.0/UDP 185.35.228.40:5080;rport;branch=z9hG4bKXX0HNjFSB1D3c. >> Max-Forwards: 69. >> From: "sipp" ;tag=eFUjHeNKv5KNg. >> To: . >> Call-ID: c755e7c8-cf13-1235-25a9-363165383663. >> CSeq: 108575420 INVITE. >> Contact: . >> User-Agent: FreeSWITCH-mod_sofia/1.6.18+git~20170612T211449Z~6e79667c0a~ >> 64bit. >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, >> REGISTER, REFER, NOTIFY. >> Supported: timer, path, replaces. >> Allow-Events: talk, hold, conference, refer. >> Content-Type: application/sdp. >> Content-Disposition: session. >> Content-Length: 222. >> X-FS-Support: update_display,send_info. >> Remote-Party-ID: "sipp" ;party >> =calling;screen=yes;privacy=off. >> . >> v=0. >> o=FreeSWITCH 1497801741 1497801743 IN IP4 185.35.228.40. >> s=FreeSWITCH. >> c=IN IP4 185.35.228.40. >> t=0 0. >> m=audio 21228 RTP/AVP 0 101. >> a=rtpmap:0 PCMU/8000. >> a=rtpmap:101 telephone-event/8000. >> a=fmtp:101 0-16. >> a=ptime:20. >> >> # >> U 185.35.228.48:5060 -> 185.35.228.40:5080 >> SIP/2.0 180 Ringing. >> Via: SIP/2.0/UDP 185.35.228.40:5080;rport;branch=z9hG4bKXX0HNjFSB1D3c. >> From: "sipp" ;tag=eFUjHeNKv5KNg. >> To: ;tag=31480SIPpTag018. >> Call-ID: c755e7c8-cf13-1235-25a9-363165383663. >> CSeq: 108575420 INVITE. >> Contact: . >> Content-Length: 0. >> . >> >> # >> U 185.35.228.48:5060 -> 185.35.228.40:5080 >> SIP/2.0 200 OK. >> Via: SIP/2.0/UDP 185.35.228.40:5080;rport;branch=z9hG4bKXX0HNjFSB1D3c. >> From: "sipp" ;tag=eFUjHeNKv5KNg. >> To: ;tag=31480SIPpTag018. >> Call-ID: c755e7c8-cf13-1235-25a9-363165383663. >> CSeq: 108575420 INVITE. >> Contact: . >> Content-Type: application/sdp. >> Content-Length: 137. >> . >> v=0. >> o=user1 53655765 2353687637 IN IP4 185.35.228.48. >> s=-. >> c=IN IP4 185.35.228.48. >> t=0 0. >> m=audio 6000 RTP/AVP 0. >> a=rtpmap:0 PCMU/8000. >> >> # >> U 185.35.228.40:5080 -> 185.35.228.48:5060 >> ACK sip:185.35.228.48:5060;transport=UDP SIP/2.0. >> Via: SIP/2.0/UDP 185.35.228.40:5080;rport;branch=z9hG4bKy6SaQD0v893Nr. >> Max-Forwards: 70. >> From: "sipp" ;tag=eFUjHeNKv5KNg. >> To: ;tag=31480SIPpTag018. >> Call-ID: c755e7c8-cf13-1235-25a9-363165383663. >> CSeq: 108575420 ACK. >> Contact: . >> Content-Length: 0. >> . >> >> # >> U 185.35.228.40:5080 -> 185.35.228.48:5060 >> BYE sip:185.35.228.48:5060;transport=UDP SIP/2.0. >> Via: SIP/2.0/UDP 185.35.228.40:5080;rport;branch=z9hG4bKZFK3r8g05jt8K. >> Max-Forwards: 70. >> From: "sipp" ;tag=eFUjHeNKv5KNg. >> To: ;tag=31480SIPpTag018. >> Call-ID: c755e7c8-cf13-1235-25a9-363165383663. >> CSeq: 108575421 BYE. >> User-Agent: FreeSWITCH-mod_sofia/1.6.18+git~20170612T211449Z~6e79667c0a~ >> 64bit. >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, >> REGISTER, REFER, NOTIFY. >> Supported: timer, path, replaces. >> Reason: SIP;cause=487;text="ORIGINATOR_CANCEL". >> Content-Length: 0. >> . >> >> # >> U 185.35.228.48:5060 -> 185.35.228.40:5080 >> SIP/2.0 200 OK. >> Via: SIP/2.0/UDP 185.35.228.40:5080;rport;branch=z9hG4bKZFK3r8g05jt8K. >> From: "sipp" ;tag=eFUjHeNKv5KNg. >> To: ;tag=31480SIPpTag018. >> Call-ID: c755e7c8-cf13-1235-25a9-363165383663. >> CSeq: 108575421 BYE. >> Contact: . >> Content-Length: 0. >> . >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From joelists at tm.net.uk Mon Jun 19 07:42:55 2017 From: joelists at tm.net.uk (Joseph Waite) Date: Mon, 19 Jun 2017 08:42:55 +0100 Subject: [Freeswitch-users] Call Dropping In-Reply-To: References: <222F1F16-373C-484A-B290-200CB13F2CAD@tm.net.uk> Message-ID: <4CE385F8-7E96-4565-AB5D-94C8459EDAD7@tm.net.uk> Ok, I had set auth_calls to false in my sip profile, however because it was not sending the Radius auth it was not getting a value for h323-credit-time. Trouble is if I set auth_calls to true then it sends a “SIP/2.0 407 Proxy Authentication Required” in response to an INVITE which I don’t want on my IP authenticated calls port. How do I get FS to not send this but still send the Radius Auth packet based simply on the IP address? Regards > On 19 Jun 2017, at 00:33, Ahmed Sboor wrote: > > if on VCS , Rate table is set and balance or credit limit is also positive and rate exist in rate table , then h323-credit-limit is also set. > you should also post mod radius debug logs. > > > On Mon, Jun 19, 2017 at 4:08 AM, Colin Morelli > wrote: > You've got an execute_on_answer of sched_hangup(+${h323-credit-time} alloted_timeout) > > Immediately after your call is answered: > 2017-06-18 22:26:34.279530 [NOTICE] switch_ivr_originate.c:3647 Channel [sofia/external/sipp at 185.35.228.51 :5060] has been answered > EXECUTE sofia/external/sipp at 185.35.228.51 :5060 sched_hangup(+ alloted_timeout) > > It would seem that h323-credit-time is not being set, which is causing sched_hangup to immediately hangup the call on answer. > > On Sun, Jun 18, 2017 at 6:52 PM, Joseph Waite > wrote: > Hi Guys > > Using FreeSwitch with Radius linked to JeraSoft VCS billing system. > > I am sending a Call from a SIPP originator, through the FreeSwitch box and back out to another SIPP terminator scenario. > The call goes through ok, everything happens as it should, however the call immediately drops, I have done egrep’s of both sides of the call and the BYE is defiantly coming from Freeswitch for some reason but I cannot work out why. Anyone any ideas? > I am attaching the FreeSwitch logs plus the egrep’s > If I register zipper on my laptop to FS and make a call works fine. > > 2017-06-18 22:26:34.199519 [NOTICE] switch_channel.c:1104 New Channel sofia/external/sipp at 185.35.228.51 :5060 [15021010-8f64-439f-8dbb-1afe090c44a5] > 2017-06-18 22:26:34.199519 [DEBUG] switch_core_state_machine.c:584 (sofia/external/sipp at 185.35.228.51 :5060) Running State Change CS_NEW (Cur 1 Tot 67) > 2017-06-18 22:26:34.199519 [DEBUG] sofia.c:9837 sofia/external/sipp at 185.35.228.51 :5060 receiving invite from 185.35.228.51:5060 version: 1.6.18 git 6e79667 2017-06-12 21:14:49Z 64bit > 2017-06-18 22:26:34.219492 [DEBUG] sofia.c:7048 Channel sofia/external/sipp at 185.35.228.51 :5060 entering state [received][100] > 2017-06-18 22:26:34.219492 [DEBUG] sofia.c:7058 Remote SDP: > v=0 > o=user1 53655765 2353687637 IN IP4 185.35.228.51 > s=- > c=IN IP4 185.35.228.51 > t=0 0 > m=audio 6000 RTP/AVP 0 > a=rtpmap:0 PCMU/8000 > > 2017-06-18 22:26:34.219492 [DEBUG] sofia.c:7450 (sofia/external/sipp at 185.35.228.51 :5060) State Change CS_NEW -> CS_INIT > 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:603 (sofia/external/sipp at 185.35.228.51 :5060) State NEW > 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:584 (sofia/external/sipp at 185.35.228.51 :5060) Running State Change CS_INIT (Cur 1 Tot 67) > 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:627 (sofia/external/sipp at 185.35.228.51 :5060) State INIT > 2017-06-18 22:26:34.219492 [DEBUG] mod_sofia.c:90 sofia/external/sipp at 185.35.228.51 :5060 SOFIA INIT > 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:40 sofia/external/sipp at 185.35.228.51 :5060 Standard INIT > 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:48 (sofia/external/sipp at 185.35.228.51 :5060) State Change CS_INIT -> CS_ROUTING > 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:627 (sofia/external/sipp at 185.35.228.51 :5060) State INIT going to sleep > 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:584 (sofia/external/sipp at 185.35.228.51 :5060) Running State Change CS_ROUTING (Cur 1 Tot 67) > 2017-06-18 22:26:34.219492 [DEBUG] switch_channel.c:2249 (sofia/external/sipp at 185.35.228.51 :5060) Callstate Change DOWN -> RINGING > 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:643 (sofia/external/sipp at 185.35.228.51 :5060) State ROUTING > 2017-06-18 22:26:34.219492 [DEBUG] mod_sofia.c:143 sofia/external/sipp at 185.35.228.51 :5060 SOFIA ROUTING > 2017-06-18 22:26:34.219492 [ERR] mod_xml_radius.c:933 Result of true match: 185.35.228.40 == ^185\.35\.229\.30 > 2017-06-18 22:26:34.219492 [INFO] mod_xml_radius.c:986 mod_xml_radius: Accounting Start success > 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:236 sofia/external/sipp at 185.35.228.51 :5060 Standard ROUTING > 2017-06-18 22:26:34.219492 [INFO] mod_dialplan_xml.c:637 Processing sipp ->441554555666 in context public > Dialplan: sofia/external/sipp at 185.35.228.51 :5060 parsing [public->unloop] continue=false > Dialplan: sofia/external/sipp at 185.35.228.51 :5060 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false > Dialplan: sofia/external/sipp at 185.35.228.51 :5060 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false > Dialplan: sofia/external/sipp at 185.35.228.51 :5060 parsing [public->outside_call] continue=true > Dialplan: sofia/external/sipp at 185.35.228.51 :5060 Absolute Condition [outside_call] > Dialplan: sofia/external/sipp at 185.35.228.51 :5060 Action set(outside_call=true) > Dialplan: sofia/external/sipp at 185.35.228.51 :5060 Action export(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) > Dialplan: sofia/external/sipp at 185.35.228.51 :5060 parsing [public->call_debug] continue=true > Dialplan: sofia/external/sipp at 185.35.228.51 :5060 Regex (FAIL) [call_debug] ${call_debug}(false) =~ /^true$/ break=never > Dialplan: sofia/external/sipp at 185.35.228.51 :5060 parsing [public->rejections] continue=false > Dialplan: sofia/external/sipp at 185.35.228.51 :5060 Regex (FAIL) [rejections] ${radius_auth_result}() =~ /2/ break=on-false > Dialplan: sofia/external/sipp at 185.35.228.51 :5060 parsing [public->timedouts] continue=false > Dialplan: sofia/external/sipp at 185.35.228.51 :5060 Regex (FAIL) [timedouts] ${radius_auth_result}() =~ /1/ break=on-false > Dialplan: sofia/external/sipp at 185.35.228.51 :5060 parsing [public->JeraSoft VCS Routing] continue=false > Dialplan: sofia/external/sipp at 185.35.228.51 :5060 Regex (PASS) [JeraSoft VCS Routing] destination_number(441554555666) =~ /^(.+)$/ break=on-false > Dialplan: sofia/external/sipp at 185.35.228.51 :5060 Action export(nolocal:h323-call-origin=originate) > Dialplan: sofia/external/sipp at 185.35.228.51 :5060 Action set(sip_h_X-accountcode=${accountcode}) > Dialplan: sofia/external/sipp at 185.35.228.51 :5060 Action set(call_direction=outbound) > Dialplan: sofia/external/sipp at 185.35.228.51 :5060 Action set(hangup_after_bridge=true) > Dialplan: sofia/external/sipp at 185.35.228.51 :5060 Action set(continue_on_fail=true) > Dialplan: sofia/external/sipp at 185.35.228.51 :5060 Action set(inherit_codec=true) > Dialplan: sofia/external/sipp at 185.35.228.51 :5060 Action set(call_timeout=20) > Dialplan: sofia/external/sipp at 185.35.228.51 :5060 Action set(fail_on_single_reject=USER_BUSY) > Dialplan: sofia/external/sipp at 185.35.228.51 :5060 Action set(origination_caller_id_name=${sip_req_user}) > Dialplan: sofia/external/sipp at 185.35.228.51 :5060 Action set(origination_caller_id_number=${sip_from_user}) > Dialplan: sofia/external/sipp at 185.35.228.51 :5060 Action set(execute_on_answer=sched_hangup +${h323-credit-time} alloted_timeout) > Dialplan: sofia/external/sipp at 185.35.228.51 :5060 Action bridge({sip_invite_from_uri=sip:${sip_from_user}@${sip_network_ip}}sofia/external/${destination_number}@185.35.229.30:5060 ) > Dialplan: sofia/external/sipp at 185.35.228.51 :5060 Action hangup(${bridge_hangup_cause}) > 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:286 (sofia/external/sipp at 185.35.228.51 :5060) State Change CS_ROUTING -> CS_EXECUTE > 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:643 (sofia/external/sipp at 185.35.228.51 :5060) State ROUTING going to sleep > 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:584 (sofia/external/sipp at 185.35.228.51 :5060) Running State Change CS_EXECUTE (Cur 1 Tot 67) > 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:650 (sofia/external/sipp at 185.35.228.51 :5060) State EXECUTE > 2017-06-18 22:26:34.219492 [DEBUG] mod_sofia.c:198 sofia/external/sipp at 185.35.228.51 :5060 SOFIA EXECUTE > 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:328 sofia/external/sipp at 185.35.228.51 :5060 Standard EXECUTE > EXECUTE sofia/external/sipp at 185.35.228.51 :5060 set(outside_call=true) > 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET sofia/external/sipp at 185.35.228.51 :5060 [outside_call]=[true] > EXECUTE sofia/external/sipp at 185.35.228.51 :5060 export(RFC2822_DATE=Sun, 18 Jun 2017 22:26:34 +0100) > 2017-06-18 22:26:34.219492 [DEBUG] switch_channel.c:1296 EXPORT (export_vars) [RFC2822_DATE]=[Sun, 18 Jun 2017 22:26:34 +0100] > EXECUTE sofia/external/sipp at 185.35.228.51 :5060 export(nolocal:h323-call-origin=originate) > 2017-06-18 22:26:34.219492 [DEBUG] switch_channel.c:1296 EXPORT (export_vars) (REMOTE ONLY) [h323-call-origin]=[originate] > EXECUTE sofia/external/sipp at 185.35.228.51 :5060 set(sip_h_X-accountcode=) > 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET sofia/external/sipp at 185.35.228.51 :5060 [sip_h_X-accountcode]=[UNDEF] > EXECUTE sofia/external/sipp at 185.35.228.51 :5060 set(call_direction=outbound) > 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET sofia/external/sipp at 185.35.228.51 :5060 [call_direction]=[outbound] > EXECUTE sofia/external/sipp at 185.35.228.51 :5060 set(hangup_after_bridge=true) > 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET sofia/external/sipp at 185.35.228.51 :5060 [hangup_after_bridge]=[true] > EXECUTE sofia/external/sipp at 185.35.228.51 :5060 set(continue_on_fail=true) > 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET sofia/external/sipp at 185.35.228.51 :5060 [continue_on_fail]=[true] > EXECUTE sofia/external/sipp at 185.35.228.51 :5060 set(inherit_codec=true) > 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET sofia/external/sipp at 185.35.228.51 :5060 [inherit_codec]=[true] > EXECUTE sofia/external/sipp at 185.35.228.51 :5060 set(call_timeout=20) > 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET sofia/external/sipp at 185.35.228.51 :5060 [call_timeout]=[20] > EXECUTE sofia/external/sipp at 185.35.228.51 :5060 set(fail_on_single_reject=USER_BUSY) > 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET sofia/external/sipp at 185.35.228.51 :5060 [fail_on_single_reject]=[USER_BUSY] > EXECUTE sofia/external/sipp at 185.35.228.51 :5060 set(origination_caller_id_name=441554555666) > 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET sofia/external/sipp at 185.35.228.51 :5060 [origination_caller_id_name]=[441554555666] > EXECUTE sofia/external/sipp at 185.35.228.51 :5060 set(origination_caller_id_number=sipp) > 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET sofia/external/sipp at 185.35.228.51 :5060 [origination_caller_id_number]=[sipp] > EXECUTE sofia/external/sipp at 185.35.228.51 :5060 set(execute_on_answer=sched_hangup + alloted_timeout) > 2017-06-18 22:26:34.219492 [DEBUG] mod_dptools.c:1530 SET sofia/external/sipp at 185.35.228.51 :5060 [execute_on_answer]=[sched_hangup + alloted_timeout] > EXECUTE sofia/external/sipp at 185.35.228.51 :5060 bridge({sip_invite_from_uri=sip:sipp at 185.35.228.51 }sofia/external/441554555666 at 185.35.229.30 :5060) > 2017-06-18 22:26:34.219492 [DEBUG] switch_channel.c:1250 sofia/external/sipp at 185.35.228.51 :5060 EXPORTING[export_vars] [RFC2822_DATE]=[Sun, 18 Jun 2017 22:26:34 +0100] to event > 2017-06-18 22:26:34.219492 [DEBUG] switch_channel.c:1250 sofia/external/sipp at 185.35.228.51 :5060 EXPORTING[export_vars] [h323-call-origin]=[originate] to event > 2017-06-18 22:26:34.219492 [DEBUG] switch_ivr_originate.c:2142 Parsing global variables > 2017-06-18 22:26:34.219492 [NOTICE] switch_channel.c:1104 New Channel sofia/external/441554555666 at 185.35.229.30 :5060 [96c1a021-5195-41ce-b903-08b98816d70d] > 2017-06-18 22:26:34.219492 [DEBUG] mod_sofia.c:4819 (sofia/external/441554555666 at 185.35.229.30 :5060) State Change CS_NEW -> CS_INIT > 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:584 (sofia/external/441554555666 at 185.35.229.30 :5060) Running State Change CS_INIT (Cur 2 Tot 68) > 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:627 (sofia/external/441554555666 at 185.35.229.30 :5060) State INIT > 2017-06-18 22:26:34.219492 [DEBUG] mod_sofia.c:90 sofia/external/441554555666 at 185.35.229.30 :5060 SOFIA INIT > 2017-06-18 22:26:34.219492 [DEBUG] sofia_glue.c:1295 sofia/external/441554555666 at 185.35.229.30 :5060 sending invite version: 1.6.18 git 6e79667 2017-06-12 21:14:49Z 64bit > Local SDP: > v=0 > o=FreeSWITCH 1497789362 1497789363 IN IP4 185.35.228.40 > s=FreeSWITCH > c=IN IP4 185.35.228.40 > t=0 0 > m=audio 31832 RTP/AVP 0 101 13 > a=rtpmap:0 PCMU/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=rtpmap:13 CN/8000 > a=ptime:20 > a=sendrecv > > 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:40 sofia/external/441554555666 at 185.35.229.30 :5060 Standard INIT > 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:48 (sofia/external/441554555666 at 185.35.229.30 :5060) State Change CS_INIT -> CS_ROUTING > 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:627 (sofia/external/441554555666 at 185.35.229.30 :5060) State INIT going to sleep > 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:584 (sofia/external/441554555666 at 185.35.229.30 :5060) Running State Change CS_ROUTING (Cur 2 Tot 68) > 2017-06-18 22:26:34.219492 [DEBUG] sofia.c:7048 Channel sofia/external/441554555666 at 185.35.229.30 :5060 entering state [calling][0] > 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:643 (sofia/external/441554555666 at 185.35.229.30 :5060) State ROUTING > 2017-06-18 22:26:34.219492 [DEBUG] mod_sofia.c:143 sofia/external/441554555666 at 185.35.229.30 :5060 SOFIA ROUTING > 2017-06-18 22:26:34.219492 [DEBUG] switch_ivr_originate.c:67 (sofia/external/441554555666 at 185.35.229.30 :5060) State Change CS_ROUTING -> CS_CONSUME_MEDIA > 2017-06-18 22:26:34.219492 [ERR] mod_xml_radius.c:930 Didn't match: 185.35.229.30:5060 == ^185\.35\.229\.30 > 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:643 (sofia/external/441554555666 at 185.35.229.30 :5060) State ROUTING going to sleep > 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:584 (sofia/external/441554555666 at 185.35.229.30 :5060) Running State Change CS_CONSUME_MEDIA (Cur 2 Tot 68) > 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:662 (sofia/external/441554555666 at 185.35.229.30 :5060) State CONSUME_MEDIA > 2017-06-18 22:26:34.219492 [DEBUG] switch_core_state_machine.c:662 (sofia/external/441554555666 at 185.35.229.30 :5060) State CONSUME_MEDIA going to sleep > 2017-06-18 22:26:34.259480 [DEBUG] sofia_glue.c:1295 sofia/external/441554555666 at 185.35.229.30 :5060 sending invite version: 1.6.18 git 6e79667 2017-06-12 21:14:49Z 64bit > Local SDP: > v=0 > o=FreeSWITCH 1497789362 1497789364 IN IP4 185.35.228.40 > s=FreeSWITCH > c=IN IP4 185.35.228.40 > t=0 0 > m=audio 31832 RTP/AVP 0 101 > a=rtpmap:0 PCMU/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > a=sendrecv > > 2017-06-18 22:26:34.259480 [DEBUG] sofia.c:7048 Channel sofia/external/441554555666 at 185.35.229.30 :5060 entering state [calling][0] > 2017-06-18 22:26:34.259480 [DEBUG] sofia.c:7048 Channel sofia/external/441554555666 at 185.35.229.30 :5060 entering state [proceeding][180] > 2017-06-18 22:26:34.259480 [NOTICE] sofia.c:7156 Ring-Ready sofia/external/441554555666 at 185.35.229.30 :5060! > 2017-06-18 22:26:34.259480 [DEBUG] switch_channel.c:3346 (sofia/external/441554555666 at 185.35.229.30 :5060) Callstate Change DOWN -> RINGING > 2017-06-18 22:26:34.259480 [DEBUG] sofia.c:7048 Channel sofia/external/441554555666 at 185.35.229.30 :5060 entering state [completing][200] > 2017-06-18 22:26:34.259480 [DEBUG] sofia.c:7058 Remote SDP: > v=0 > o=user1 53655765 2353687637 IN IP4 185.35.228.48 > s=- > c=IN IP4 185.35.228.48 > t=0 0 > m=audio 6000 RTP/AVP 0 > a=rtpmap:0 PCMU/8000 > > 2017-06-18 22:26:34.259480 [DEBUG] sofia.c:7048 Channel sofia/external/441554555666 at 185.35.229.30 :5060 entering state [ready][200] > 2017-06-18 22:26:34.259480 [DEBUG] switch_core_media.c:4445 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] > 2017-06-18 22:26:34.259480 [DEBUG] switch_core_media.c:4500 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match > 2017-06-18 22:26:34.259480 [DEBUG] switch_core_media.c:3057 Set Codec sofia/external/441554555666 at 185.35.229.30 :5060 PCMU/8000 20 ms 160 samples 64000 bits 1 channels > 2017-06-18 22:26:34.259480 [DEBUG] switch_core_codec.c:111 sofia/external/441554555666 at 185.35.229.30 :5060 Original read codec set to PCMU:0 > 2017-06-18 22:26:34.259480 [DEBUG] switch_core_media.c:4770 No 2833 in SDP. Disable 2833 dtmf and switch to INFO > 2017-06-18 22:26:34.259480 [DEBUG] switch_core_media.c:6874 AUDIO RTP [sofia/external/441554555666 at 185.35.229.30 :5060] 185.35.228.40 port 31832 -> 185.35.228.48 port 6000 codec: 0 ms: 20 > 2017-06-18 22:26:34.259480 [DEBUG] switch_rtp.c:4108 Starting timer [soft] 160 bytes per 20ms > 2017-06-18 22:26:34.279530 [DEBUG] switch_core_media.c:7205 sofia/external/441554555666 at 185.35.229.30 :5060 Set rtp dtmf delay to 40 > 2017-06-18 22:26:34.279530 [NOTICE] sofia.c:8182 Channel [sofia/external/441554555666 at 185.35.229.30 :5060] has been answered > 2017-06-18 22:26:34.279530 [DEBUG] switch_channel.c:3773 (sofia/external/441554555666 at 185.35.229.30 :5060) Callstate Change RINGING -> ACTIVE > 2017-06-18 22:26:34.279530 [NOTICE] mod_sofia.c:2273 Ring-Ready sofia/external/sipp at 185.35.228.51 :5060! > 2017-06-18 22:26:34.279530 [DEBUG] sofia.c:7048 Channel sofia/external/sipp at 185.35.228.51 :5060 entering state [early][180] > 2017-06-18 22:26:34.279530 [NOTICE] switch_ivr_originate.c:525 Ring Ready sofia/external/sipp at 185.35.228.51 :5060! > 2017-06-18 22:26:34.279530 [DEBUG] switch_ivr_originate.c:410 Setting codec string on sofia/external/sipp at 185.35.228.51 :5060 to PCMU at 8000h@20i > 2017-06-18 22:26:34.279530 [DEBUG] switch_core_media.c:4445 Audio Codec Compare [PCMU:0:8000:20:64000:1]/[PCMU:0:8000:20:64000:1] > 2017-06-18 22:26:34.279530 [DEBUG] switch_core_media.c:4500 Audio Codec Compare [PCMU:0:8000:20:64000:1] ++++ is saved as a match > 2017-06-18 22:26:34.279530 [DEBUG] switch_core_media.c:3057 Set Codec sofia/external/sipp at 185.35.228.51 :5060 PCMU/8000 20 ms 160 samples 64000 bits 1 channels > 2017-06-18 22:26:34.279530 [DEBUG] switch_core_codec.c:111 sofia/external/sipp at 185.35.228.51 :5060 Original read codec set to PCMU:0 > 2017-06-18 22:26:34.279530 [DEBUG] switch_core_media.c:4770 No 2833 in SDP. Disable 2833 dtmf and switch to INFO > 2017-06-18 22:26:34.279530 [DEBUG] switch_core_media.c:6874 AUDIO RTP [sofia/external/sipp at 185.35.228.51 :5060] 185.35.228.40 port 23728 -> 185.35.228.51 port 6000 codec: 0 ms: 20 > 2017-06-18 22:26:34.279530 [DEBUG] switch_rtp.c:4108 Starting timer [soft] 160 bytes per 20ms > 2017-06-18 22:26:34.279530 [DEBUG] switch_core_media.c:7205 sofia/external/sipp at 185.35.228.51 :5060 Set rtp dtmf delay to 40 > 2017-06-18 22:26:34.279530 [NOTICE] sofia_media.c:92 Pre-Answer sofia/external/sipp at 185.35.228.51 :5060! > 2017-06-18 22:26:34.279530 [DEBUG] switch_channel.c:3474 (sofia/external/sipp at 185.35.228.51 :5060) Callstate Change RINGING -> EARLY > 2017-06-18 22:26:34.279530 [DEBUG] switch_core_media.c:6857 Audio params are unchanged for sofia/external/sipp at 185.35.228.51 :5060. > 2017-06-18 22:26:34.279530 [DEBUG] mod_sofia.c:850 Local SDP sofia/external/sipp at 185.35.228.51 :5060: > v=0 > o=FreeSWITCH 1497797466 1497797467 IN IP4 185.35.228.40 > s=FreeSWITCH > c=IN IP4 185.35.228.40 > t=0 0 > m=audio 23728 RTP/AVP 0 > a=rtpmap:0 PCMU/8000 > a=ptime:20 > a=sendrecv > > 2017-06-18 22:26:34.279530 [NOTICE] switch_ivr_originate.c:3647 Channel [sofia/external/sipp at 185.35.228.51 :5060] has been answered > EXECUTE sofia/external/sipp at 185.35.228.51 :5060 sched_hangup(+ alloted_timeout) > 2017-06-18 22:26:34.279530 [DEBUG] sofia.c:7048 Channel sofia/external/sipp at 185.35.228.51 :5060 entering state [completed][200] > 2017-06-18 22:26:34.279530 [DEBUG] sofia.c:7048 Channel sofia/external/sipp at 185.35.228.51 :5060 entering state [ready][200] > 2017-06-18 22:26:34.279530 [NOTICE] mod_dptools.c:1188 Hangup sofia/external/sipp at 185.35.228.51 :5060 [CS_EXECUTE] [NORMAL_CLEARING] > 2017-06-18 22:26:34.279530 [DEBUG] switch_core_session.c:2814 sofia/external/sipp at 185.35.228.51 :5060 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) > 2017-06-18 22:26:34.279530 [DEBUG] switch_channel.c:3773 (sofia/external/sipp at 185.35.228.51 :5060) Callstate Change EARLY -> ACTIVE > 2017-06-18 22:26:34.279530 [DEBUG] switch_ivr_originate.c:3647 sofia/external/sipp at 185.35.228.51 :5060 skip receive message [ANSWER_EVENT] (channel is hungup already) > 2017-06-18 22:26:34.319525 [DEBUG] switch_ivr_originate.c:3848 Originate Resulted in Error Cause: 487 [ORIGINATOR_CANCEL] > 2017-06-18 22:26:34.319525 [NOTICE] switch_ivr_originate.c:3938 Hangup sofia/external/441554555666 at 185.35.229.30 :5060 [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL] > 2017-06-18 22:26:34.319525 [DEBUG] switch_core_state_machine.c:584 (sofia/external/441554555666 at 185.35.229.30 :5060) Running State Change CS_HANGUP (Cur 2 Tot 68) > 2017-06-18 22:26:34.319525 [DEBUG] switch_core_state_machine.c:850 (sofia/external/441554555666 at 185.35.229.30 :5060) Callstate Change ACTIVE -> HANGUP > 2017-06-18 22:26:34.319525 [DEBUG] switch_core_state_machine.c:852 (sofia/external/441554555666 at 185.35.229.30 :5060) State HANGUP > 2017-06-18 22:26:34.319525 [DEBUG] mod_sofia.c:438 Channel sofia/external/441554555666 at 185.35.229.30 :5060 hanging up, cause: ORIGINATOR_CANCEL > 2017-06-18 22:26:34.319525 [INFO] mod_dptools.c:3418 Originate Failed. Cause: ORIGINATOR_CANCEL > 2017-06-18 22:26:34.319525 [DEBUG] mod_sofia.c:491 Sending BYE to sofia/external/441554555666 at 185.35.229.30 :5060 > 2017-06-18 22:26:34.319525 [DEBUG] switch_core_state_machine.c:60 sofia/external/441554555666 at 185.35.229.30 :5060 Standard HANGUP, cause: ORIGINATOR_CANCEL > 2017-06-18 22:26:34.319525 [DEBUG] switch_core_state_machine.c:852 (sofia/external/441554555666 at 185.35.229.30 :5060) State HANGUP going to sleep > 2017-06-18 22:26:34.319525 [DEBUG] switch_core_state_machine.c:619 (sofia/external/441554555666 at 185.35.229.30 :5060) State Change CS_HANGUP -> CS_REPORTING > 2017-06-18 22:26:34.319525 [DEBUG] switch_core_session.c:2814 sofia/external/sipp at 185.35.228.51 :5060 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) > 2017-06-18 22:26:34.319525 [DEBUG] switch_core_state_machine.c:584 (sofia/external/441554555666 at 185.35.229.30 :5060) Running State Change CS_REPORTING (Cur 2 Tot 68) > 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:650 (sofia/external/sipp at 185.35.228.51 :5060) State EXECUTE going to sleep > 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:584 (sofia/external/sipp at 185.35.228.51 :5060) Running State Change CS_HANGUP (Cur 2 Tot 68) > 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:938 (sofia/external/441554555666 at 185.35.229.30 :5060) State REPORTING > 2017-06-18 22:26:34.339569 [ERR] mod_xml_radius.c:930 Didn't match: 185.35.229.30 == ^185\.35\.229\.30 > 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:850 (sofia/external/sipp at 185.35.228.51 :5060) Callstate Change ACTIVE -> HANGUP > 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:174 sofia/external/441554555666 at 185.35.229.30 :5060 Standard REPORTING, cause: ORIGINATOR_CANCEL > 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:938 (sofia/external/441554555666 at 185.35.229.30 :5060) State REPORTING going to sleep > 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:852 (sofia/external/sipp at 185.35.228.51 :5060) State HANGUP > 2017-06-18 22:26:34.339569 [DEBUG] mod_sofia.c:438 Channel sofia/external/sipp at 185.35.228.51 :5060 hanging up, cause: NORMAL_CLEARING > 2017-06-18 22:26:34.339569 [DEBUG] mod_sofia.c:491 Sending BYE to sofia/external/sipp at 185.35.228.51 :5060 > 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:60 sofia/external/sipp at 185.35.228.51 :5060 Standard HANGUP, cause: NORMAL_CLEARING > 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:852 (sofia/external/sipp at 185.35.228.51 :5060) State HANGUP going to sleep > 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:619 (sofia/external/sipp at 185.35.228.51 :5060) State Change CS_HANGUP -> CS_REPORTING > 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:584 (sofia/external/sipp at 185.35.228.51 :5060) Running State Change CS_REPORTING (Cur 2 Tot 68) > 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:938 (sofia/external/sipp at 185.35.228.51 :5060) State REPORTING > 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:610 (sofia/external/441554555666 at 185.35.229.30 :5060) State Change CS_REPORTING -> CS_DESTROY > 2017-06-18 22:26:34.339569 [DEBUG] switch_core_session.c:1664 Session 68 (sofia/external/441554555666 at 185.35.229.30 :5060) Locked, Waiting on external entities > 2017-06-18 22:26:34.339569 [ERR] mod_xml_radius.c:933 Result of true match: 185.35.228.40 == ^185\.35\.229\.30 > 2017-06-18 22:26:34.339569 [NOTICE] switch_core_session.c:1682 Session 68 (sofia/external/441554555666 at 185.35.229.30 :5060) Ended > 2017-06-18 22:26:34.339569 [NOTICE] switch_core_session.c:1686 Close Channel sofia/external/441554555666 at 185.35.229.30 :5060 [CS_DESTROY] > 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:741 (sofia/external/441554555666 at 185.35.229.30 :5060) Running State Change CS_DESTROY (Cur 1 Tot 68) > 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:751 (sofia/external/441554555666 at 185.35.229.30 :5060) State DESTROY > 2017-06-18 22:26:34.339569 [DEBUG] mod_sofia.c:343 sofia/external/441554555666 at 185.35.229.30 :5060 SOFIA DESTROY > 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:181 sofia/external/441554555666 at 185.35.229.30 :5060 Standard DESTROY > 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:751 (sofia/external/441554555666 at 185.35.229.30 :5060) State DESTROY going to sleep > 2017-06-18 22:26:34.339569 [INFO] mod_xml_radius.c:1044 mod_xml_radius: Accounting Stop success > 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:174 sofia/external/sipp at 185.35.228.51 :5060 Standard REPORTING, cause: NORMAL_CLEARING > 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:938 (sofia/external/sipp at 185.35.228.51 :5060) State REPORTING going to sleep > 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:610 (sofia/external/sipp at 185.35.228.51 :5060) State Change CS_REPORTING -> CS_DESTROY > 2017-06-18 22:26:34.339569 [DEBUG] switch_core_session.c:1664 Session 67 (sofia/external/sipp at 185.35.228.51 :5060) Locked, Waiting on external entiti > 2017-06-18 22:26:34.339569 [NOTICE] switch_core_session.c:1682 Session 67 (sofia/external/sipp at 185.35.228.51 :5060) Ended > 2017-06-18 22:26:34.339569 [NOTICE] switch_core_session.c:1686 Close Channel sofia/external/sipp at 185.35.228.51 :5060 [CS_DESTROY] > 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:741 (sofia/external/sipp at 185.35.228.51 :5060) Running State Change CS_DESTROY (Cur 0 Tot 68) > 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:751 (sofia/external/sipp at 185.35.228.51 :5060) State DESTROY > 2017-06-18 22:26:34.339569 [DEBUG] mod_sofia.c:343 sofia/external/sipp at 185.35.228.51 :5060 SOFIA DESTROY > 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:181 sofia/external/sipp at 185.35.228.51 :5060 Standard DESTROY > 2017-06-18 22:26:34.339569 [DEBUG] switch_core_state_machine.c:751 (sofia/external/sipp at 185.35.228.51 :5060) State DESTROY going to sleep > > > U 185.35.228.51:5060 -> 185.35.228.40:5080 > INVITE sip:441554555666 at 185.35.228.40:5080 <> SIP/2.0. > Via: SIP/2.0/UDP 185.35.228.51:5060;branch=z9hG4bK-27036-1-0. > From: sipp >;tag=27036SIPpTag001. > To: 441554555666 >. > Call-ID: 1-27036 at 185.35.228.51 . > CSeq: 1 INVITE. > Contact: sip:sipp at 185.35.228.51:5060 <>. > Max-Forwards: 70. > Subject: Performance Test. > Content-Type: application/sdp. > Content-Length: 137. > . > v=0. > o=user1 53655765 2353687637 IN IP4 185.35.228.51. > s=-. > c=IN IP4 185.35.228.51. > t=0 0. > m=audio 6000 RTP/AVP 0. > a=rtpmap:0 PCMU/8000. > > # > U 185.35.228.40:5080 -> 185.35.228.51:5060 > SIP/2.0 100 Trying. > Via: SIP/2.0/UDP 185.35.228.51:5060;branch=z9hG4bK-27036-1-0. > From: sipp >;tag=27036SIPpTag001. > To: 441554555666 >. > Call-ID: 1-27036 at 185.35.228.51 . > CSeq: 1 INVITE. > User-Agent: FreeSWITCH-mod_sofia/1.6.18+git~20170612T211449Z~6e79667c0a~64bit. > Content-Length: 0. > . > > # > U 185.35.228.40:5080 -> 185.35.228.51:5060 > SIP/2.0 180 Ringing. > Via: SIP/2.0/UDP 185.35.228.51:5060;branch=z9hG4bK-27036-1-0. > From: sipp >;tag=27036SIPpTag001. > To: 441554555666 >;tag=91vp8601aS4Qp. > Call-ID: 1-27036 at 185.35.228.51 . > CSeq: 1 INVITE. > Contact: >. > User-Agent: FreeSWITCH-mod_sofia/1.6.18+git~20170612T211449Z~6e79667c0a~64bit. > Accept: application/sdp. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY. > Supported: timer, path, replaces. > Allow-Events: talk, hold, conference, refer. > Content-Length: 0. > Remote-Party-ID: "Outbound Call" >;party=calling;privacy=off;screen=no. > . > > # > U 185.35.228.40:5080 -> 185.35.228.51:5060 > SIP/2.0 200 OK. > Via: SIP/2.0/UDP 185.35.228.51:5060;branch=z9hG4bK-27036-1-0. > From: sipp >;tag=27036SIPpTag001. > To: 441554555666 >;tag=91vp8601aS4Qp. > Call-ID: 1-27036 at 185.35.228.51 . > CSeq: 1 INVITE. > Contact: >. > User-Agent: FreeSWITCH-mod_sofia/1.6.18+git~20170612T211449Z~6e79667c0a~64bit. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY. > Supported: timer, path, replaces. > Allow-Events: talk, hold, conference, refer. > Content-Type: application/sdp. > Content-Disposition: session. > Content-Length: 166. > Remote-Party-ID: "Outbound Call" >;party=calling;privacy=off;screen=no. > . > v=0. > o=FreeSWITCH 1497797499 1497797500 IN IP4 185.35.228.40. > s=FreeSWITCH. > c=IN IP4 185.35.228.40. > t=0 0. > m=audio 25252 RTP/AVP 0. > a=rtpmap:0 PCMU/8000. > a=ptime:20. > > # > U 185.35.228.51:5060 -> 185.35.228.40:5080 > ACK sip:441554555666 at 185.35.228.40:5080 <> SIP/2.0. > Via: SIP/2.0/UDP 185.35.228.51:5060;branch=z9hG4bK-27036-1-5. > From: sipp >;tag=27036SIPpTag001. > To: 441554555666 >;tag=91vp8601aS4Qp. > Call-ID: 1-27036 at 185.35.228.51 . > CSeq: 1 ACK. > Contact: sip:sipp at 185.35.228.51:5060 <>. > Max-Forwards: 70. > Subject: Performance Test. > Content-Length: 0. > . > > # > U 185.35.228.40:5080 -> 185.35.228.51:5060 > BYE sip:sipp at 185.35.228.51:5060 <> SIP/2.0. > Via: SIP/2.0/UDP 185.35.228.40:5080;rport;branch=z9hG4bKUBe0HvDjHF1XN. > Max-Forwards: 70. > From: 441554555666 >;tag=91vp8601aS4Qp. > To: sipp >;tag=27036SIPpTag001. > Call-ID: 1-27036 at 185.35.228.51 . > CSeq: 108575311 BYE. > User-Agent: FreeSWITCH-mod_sofia/1.6.18+git~20170612T211449Z~6e79667c0a~64bit. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY. > Supported: timer, path, replaces. > Reason: Q.850;cause=16;text="NORMAL_CLEARING". > Content-Length: 0. > . > > # > U 185.35.228.51:5060 -> 185.35.228.40:5080 > SIP/2.0 200 OK. > Via: SIP/2.0/UDP 185.35.228.40:5080;rport;branch=z9hG4bKUBe0HvDjHF1XN. > From: 441554555666 >;tag=91vp8601aS4Qp. > To: sipp >;tag=27036SIPpTag001. > Call-ID: 1-27036 at 185.35.228.51 . > CSeq: 108575311 BYE. > Contact: >. > Content-Length: 0. > > > > NGREP of SIP messages from FS to terminator > > > U 185.35.228.40:5080 -> 185.35.228.48:5060 > INVITE sip:441554555666 at 185.35.228.48:5060 <> SIP/2.0. > Via: SIP/2.0/UDP 185.35.228.40:5080;rport;branch=z9hG4bKXX0HNjFSB1D3c. > Max-Forwards: 69. > From: "sipp" >;tag=eFUjHeNKv5KNg. > To: >. > Call-ID: c755e7c8-cf13-1235-25a9-363165383663. > CSeq: 108575420 INVITE. > Contact: >. > User-Agent: FreeSWITCH-mod_sofia/1.6.18+git~20170612T211449Z~6e79667c0a~64bit. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY. > Supported: timer, path, replaces. > Allow-Events: talk, hold, conference, refer. > Content-Type: application/sdp. > Content-Disposition: session. > Content-Length: 222. > X-FS-Support: update_display,send_info. > Remote-Party-ID: "sipp" >;party=calling;screen=yes;privacy=off. > . > v=0. > o=FreeSWITCH 1497801741 1497801743 IN IP4 185.35.228.40. > s=FreeSWITCH. > c=IN IP4 185.35.228.40. > t=0 0. > m=audio 21228 RTP/AVP 0 101. > a=rtpmap:0 PCMU/8000. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=ptime:20. > > # > U 185.35.228.48:5060 -> 185.35.228.40:5080 > SIP/2.0 180 Ringing. > Via: SIP/2.0/UDP 185.35.228.40:5080;rport;branch=z9hG4bKXX0HNjFSB1D3c. > From: "sipp" >;tag=eFUjHeNKv5KNg. > To: >;tag=31480SIPpTag018. > Call-ID: c755e7c8-cf13-1235-25a9-363165383663. > CSeq: 108575420 INVITE. > Contact: >. > Content-Length: 0. > . > > # > U 185.35.228.48:5060 -> 185.35.228.40:5080 > SIP/2.0 200 OK. > Via: SIP/2.0/UDP 185.35.228.40:5080;rport;branch=z9hG4bKXX0HNjFSB1D3c. > From: "sipp" >;tag=eFUjHeNKv5KNg. > To: >;tag=31480SIPpTag018. > Call-ID: c755e7c8-cf13-1235-25a9-363165383663. > CSeq: 108575420 INVITE. > Contact: >. > Content-Type: application/sdp. > Content-Length: 137. > . > v=0. > o=user1 53655765 2353687637 IN IP4 185.35.228.48. > s=-. > c=IN IP4 185.35.228.48. > t=0 0. > m=audio 6000 RTP/AVP 0. > a=rtpmap:0 PCMU/8000. > > # > U 185.35.228.40:5080 -> 185.35.228.48:5060 > ACK sip:185.35.228.48:5060;transport=UDP <> SIP/2.0. > Via: SIP/2.0/UDP 185.35.228.40:5080;rport;branch=z9hG4bKy6SaQD0v893Nr. > Max-Forwards: 70. > From: "sipp" >;tag=eFUjHeNKv5KNg. > To: >;tag=31480SIPpTag018. > Call-ID: c755e7c8-cf13-1235-25a9-363165383663. > CSeq: 108575420 ACK. > Contact: >. > Content-Length: 0. > . > > # > U 185.35.228.40:5080 -> 185.35.228.48:5060 > BYE sip:185.35.228.48:5060;transport=UDP <> SIP/2.0. > Via: SIP/2.0/UDP 185.35.228.40:5080;rport;branch=z9hG4bKZFK3r8g05jt8K. > Max-Forwards: 70. > From: "sipp" >;tag=eFUjHeNKv5KNg. > To: >;tag=31480SIPpTag018. > Call-ID: c755e7c8-cf13-1235-25a9-363165383663. > CSeq: 108575421 BYE. > User-Agent: FreeSWITCH-mod_sofia/1.6.18+git~20170612T211449Z~6e79667c0a~64bit. > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY. > Supported: timer, path, replaces. > Reason: SIP;cause=487;text="ORIGINATOR_CANCEL". > Content-Length: 0. > . > > # > U 185.35.228.48:5060 -> 185.35.228.40:5080 > SIP/2.0 200 OK. > Via: SIP/2.0/UDP 185.35.228.40:5080;rport;branch=z9hG4bKZFK3r8g05jt8K. > From: "sipp" >;tag=eFUjHeNKv5KNg. > To: >;tag=31480SIPpTag018. > Call-ID: c755e7c8-cf13-1235-25a9-363165383663. > CSeq: 108575421 BYE. > Contact: >. > Content-Length: 0. > . > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From Chris.Young at enghouse.com Mon Jun 19 09:05:12 2017 From: Chris.Young at enghouse.com (Chris Young) Date: Mon, 19 Jun 2017 09:05:12 +0000 Subject: [Freeswitch-users] "Cannot Blind Transfer 1 Legged calls" message Message-ID: <95a7f732146d4cd8a3ca480e8a8e1679@UK-MAIL-001.edge.local> Hi all, I'm hoping that some of you clever folk may be able to help me out with a REFER problem I'm facing. If I originate a call from FreeSWITCH to an IVR, and the IVR attempts to transfer the call to another destination using REFER, FreeSWITCH fails with a 403 Forbidden error and reports 'Cannot Blind Transfer 1 Legged calls' in the log. FreeSWITCH is quite correct that there is only one leg of course but should it not be possible for an INVITE to be sent to the address in the Refer-To header anyway? This scenario seems more or less the same as the first 'basic transfer' example in RFC5589 so I think it should be possible unless I am missing something obvious. Are there any special configuration options needed to make this work? I tried setting proxy-refer but it didn't make any difference. Kind regards, Chris Chris Young Senior Software Engineer [cid:image7482a0.PNG at dc00f514.4fb88205] t: +44 118 943 9249 e: chris.young at enghouse.com w: www.enghouseinteractive.co.uk [cid:image6c1ba6.PNG at b8963a35.449174f5] Enghouse Interactive (UK) Ltd is a company registered in England and Wales. Registered number: 04230977. Registered office: Imperium, Imperial Way, Reading, Berkshire, RG2 0TD -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.png Type: image/png Size: 1045 bytes Desc: image001.png URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: image002.png Type: image/png Size: 5097 bytes Desc: image002.png URL: From adrian.worutowicz at esifrance.net Mon Jun 19 12:52:31 2017 From: adrian.worutowicz at esifrance.net (Adrian Worutowicz) Date: Mon, 19 Jun 2017 14:52:31 +0200 Subject: [Freeswitch-users] Build Problem in VS2015 In-Reply-To: References: <80c901d2e5f4$7e26f930$7a74eb90$@tollfreegateway.com> <594417cf.462ded0a.12ca4.f4aeSMTPIN_ADDED_BROKEN@mx.google.com> Message-ID: <008401d2e8fa$ea804ab0$bf80e010$@worutowicz@esifrance.net> I asked a friend of mine to compile and it worked fine. So it is either a pb of my VS2015 install or a question of Win7 32/64 bit (I have the 32bit version). Thanks, A. De : FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de gregor at infomedia.si -- Gregor Nanger Envoyé : vendredi 16 juin 2017 21:19 À : FreeSWITCH Users Help Objet : Re: [Freeswitch-users] Build Problem in VS2015 I do not have problems compiling with visual Studio. Except for cloning, I use same command as stated in wiki. Then open in visual Studio and Build solution. On Fri, Jun 16, 2017, 19:39 Adrian Worutowicz wrote: I followed your steps, but unfortunately I got the same result. Probably I’m missing something in my VS install. Thanks a lot anyway… De : FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] De la part de krice at tollfreegateway.com Envoyé : jeudi 15 juin 2017 18:29 À : 'FreeSWITCH Users Help' Objet : Re: [Freeswitch-users] Build Problem in VS2015 Not sure whats you are doing incorrect here, but I have just built master, I use the built in git bits with VS2015, and then drop to a command prompt (via the team explorer tab, select branches, right click the repo and select open command prompt) Then git pull, git clean -fdx, git reset –hard origin/master , git pull >From here back to the solution explorer open the FreeSWITCH.2015 solution file and build as normal… I think you have something skewed there old ssl vs new ssl bits From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Adrian Worutowicz Sent: Thursday, June 15, 2017 4:44 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Build Problem in VS2015 Hello, I try to recompile FS in VS2015 without success. I took FS sources from git master. git config --global core.autocrlf false git clone https://stash.freeswitch.org/scm/fs/freeswitch.git /c/ESI/Components/FreeSwitch/ I have wix311 for VS2015 installed. For example it searches in folder 'openssl-1.0.2k' while only a folder 'openssl' exists. I tried to recompile mod_PortAudio, and I got c:\ESI\components\freeswitch\src\mod\endpoints\mod_portaudio\pablio.h(55): fatal error C1083: Impossible d'ouvrir le fichier include : 'portaudio.h' : No such file or directory. Indeed 'portaudio.h' does not exist. Plenty of other errors in the attached file. What do I miss? Thanks in advance, Adrian. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Gregor Nanger CTO t./f.: 00386 (0) 7 6000 308/309 • m:. 00386 (0)41 756485 • Infomedia d.o.o. • Jerebova 3, Novo mesto, Slovenia • www.infomedia.si -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Mon Jun 19 16:34:56 2017 From: mike at jerris.com (Michael Jerris) Date: Mon, 19 Jun 2017 12:34:56 -0400 Subject: [Freeswitch-users] Callcenter module, can I originate call for an agent in uuid-standby mode ? In-Reply-To: References: Message-ID: <4770F7F5-BCBB-47B6-A909-263EEFEADFE4@jerris.com> mod_fifo is a much more feature rich version of a call queue than mod_callcenter. You might want to check that out instead. > On Jun 14, 2017, at 8:09 AM, Khalil Khamlichi wrote: > > Hi, > > I need to give my agents ability to make manual calls, hopefully without leaving their actually established call (they are in uuid-standby mode and in Idle state so there is no live member on the line ). > > my questions: > > Is it possible to originate a new call and bridge with agent uuid-standby session ? > would it not break the callcenter establised uuid-standby session ? > would the agent return to its uuid-standby session after the originated call is hangup ? > > ofcourse if this is too complicated, I would just connect the agent thru a second line, while leaving his uuid-standby call on the first line, though it would be so cool to somehow stay on that same uuid-standby session and enjoy both calllcenter module and manual dialing. > > Thanks in advance, and I appreciate your help. > > Khalil From mike at jerris.com Mon Jun 19 16:43:14 2017 From: mike at jerris.com (Michael Jerris) Date: Mon, 19 Jun 2017 12:43:14 -0400 Subject: [Freeswitch-users] Freeswitch sslv3 support In-Reply-To: References: Message-ID: <83A13BEB-579E-4B56-9C4B-9B0AA355A4EE@jerris.com> our websocket code should already be limited to either tls 1.1 and tls 1.2 or just tls 1.2 (i can’t remember which)… if this isn’t the case, please open a Jira for this. There should be no browsers that support web sockets that don’t support at least tls 1.1. > On Jun 15, 2017, at 4:07 AM, Agustí Ubalde Bellot wrote: > > Hi Brian, > > Is possible to disable for web socket secure connections too? > > > Thanks, > Agustí > > 2017-06-13 13:24 GMT+02:00 Agustí Ubalde Bellot >: > Hi all, > > Is there a FreeSWITCH update where sslv3 support is disabled? > > > Thanks, > Agustí -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Mon Jun 19 16:45:56 2017 From: mike at jerris.com (Michael Jerris) Date: Mon, 19 Jun 2017 12:45:56 -0400 Subject: [Freeswitch-users] question about HA solution In-Reply-To: References: <8A13E0EC-FA13-4267-80F6-CE1A8E8360CF@jerris.com> Message-ID: what is “channels_pkey” … thats not something thats anywhere in our codebase. > On Jun 15, 2017, at 5:40 AM, Igor Olhovskiy wrote: > > Hi! > Same situation here. > Idea is: > I’m having Freeswitch HA (keepalived, working, same database, calls recovering…) > If I look on «show calls» at slave node, I see calls on master node. > I crash master node (with «fsctl crash»), calls are transferred to slave node, restored, but when I run «show calls» on this (slave) node again, I see 0 calls. But calls are actually going on. > > So, it’s seems impossible to have 2nd recover on already recovered call. > > > In DB logs seen an errors like > insert into channels (uuid,direction,created,created_epoch, name,state,callstate,dialplan,context,hostname,initial_cid_name,initial_cid_num,initial_ip_addr,initial_dest,initial_dialplan,initial_context) values('57904410-a8ad-4c28-a88a-83bd2280e146','outbound','2017-06-15 19:30:32','1497519032','sofia/internal/113-akbepcb59gt2a at 172.17.240.50:5060 ','CS_INIT','DOWN','XML','sip303.empowervoice.com ','blueAPACHE_test','103','103','172.17.240.50','113-akbepcb59gt2a','XML','sip303.empowervoice.com ') > Jun 15 19:30:31 E2-EVL-T-DB-01 postgres[28042]: [109-1] 2017-06-15 19:30:31 AEST [28042-103] freeswitch at freeswitch ERROR: duplicate key value violates unique constraint «channels_pkey" > Or like > statement: insert into calls (call_uuid,call_created,call_created_epoch,caller_uuid,callee_uuid,hostname) values ('ffaf3eb5-3fc5-47fe-adef-cc4dddf53bab','2017-06-15 19:30:32','1497519032','ffaf3eb5-3fc5-47fe-adef-cc4dddf53bab','57904410-a8ad-4c28-a88a-83bd2280e146','blueAPACHE_test') > Jun 15 19:30:31 E2-EVL-T-DB-01 postgres[28042]: [147-1] 2017-06-15 19:30:31 AEST [28042-142] freeswitch at freeswitch ERROR: duplicate key value violates unique constraint «calls_pkey" > > Also I see much queries like this > delete from calls where (caller_uuid=‘ffaf3eb5-3fc5-47fe-adef-cc4dddf53bab’ or callee_uuid='ffaf3eb5-3fc5-47fe-adef-cc4dddf53bab') > delete from recovery where runtime_uuid!=‘91f571c5-e0d2-462e-aa84-e4ca07052119’ and technology=‘sofia’…. > when calls are switched. > > So, can this help to point an issue? > > 2017-06-08 18:48 GMT+03:00 Michael Jerris >: > check your db logs as nothing we are doing should be clearing those. > > On Thu, Jun 8, 2017 at 4:08 AM Denys Pozniak > wrote: > Hello! > > My configs: > > switch.conf.xml > > > > > > > > > external.conf.xml > > > > > > On 7 June 2017 at 17:35, Michael Jerris > wrote: > That param should keep it from doing so, if its not you are not setting it somehow or something else is wiping the db. > >> On Jun 5, 2017, at 1:50 PM, Denys Pozniak > wrote: >> >> Yes, correct. But when you restart FS on slave, it will erase database. And option auto-clear-sql=false not working for me. >> >> On Jun 5, 2017 6:32 PM, "Michael Jerris" > wrote: >> recovered calls will get new entries in the table. >> >>> On Jun 5, 2017, at 7:41 AM, Denys Pozniak > wrote: >>> >>> Hello! >>> >>> Thank you Raymond about your explanation, but I dont agree with some point: >>> If it really need an answer about your question -- "if it is possible to move calls back". I think it's unnecessary. - in my case I have two not equal servers, so I need to have only one as a master. >>> If switchover happens I need to have ability to restore master back. >>> >>> Thank you Luis for your link, you can do simple test to understand what I am talking about: do call -> check on master and slave #show channels -> restart FS on slave -> check on master #show channels. In my case I dont see any active calls after this, so restoring back is not possible. >>> >>> >>> >>> On 3 June 2017 at 22:16, Luis Daniel Lucio Quiroz > wrote: >>> You may want to read this article. >>> >>> http://inside-out.xyz/technology/how-to-configure-freeswitch-for-ha.html >>> >>> Le 31 mai 2017 6:29 PM, "Denys Pozniak" > a écrit : >>> Hello! >>> >>> I built FS HA solution based on keepalived and mysql master-master. >>> It works ok generally, but as I understand FS after restarting cleaning own database. >>> >>> So when node1 fails calls jump to node2, after script restarts node1 it is not possible to move calls back. >>> >>> Tried options in switch.conf.xml, but no luck: >>> >>> >>> >>> >>> Is there is a way to solve this? >>> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -- > Best regards, > Igor > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Mon Jun 19 16:46:52 2017 From: mike at jerris.com (Michael Jerris) Date: Mon, 19 Jun 2017 12:46:52 -0400 Subject: [Freeswitch-users] group_confirm_file multiple files In-Reply-To: References: Message-ID: https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools%3A+file_string > On Jun 15, 2017, at 5:54 AM, Matt Broad wrote: > > Hi, > > I'm wondering if it is possible to play multiple files using the group_confirm_file function. > > I have 2 audio files that I would like to play 1 after the other and then wait for the confirm key. > > > I have tried using mod_file_string, but get an error "Error from mpg123: File access error. (code 22)", I assume this is due to the fact it is reading the file string as one file rather than 2 separated by the ! delimiter. > -------------- next part -------------- An HTML attachment was scrubbed... URL: From michael at mailworks.org Mon Jun 19 16:48:11 2017 From: michael at mailworks.org (Michael Avers) Date: Mon, 19 Jun 2017 09:48:11 -0700 Subject: [Freeswitch-users] Callcenter module, can I originate call for an agent in uuid-standby mode ? In-Reply-To: <4770F7F5-BCBB-47B6-A909-263EEFEADFE4@jerris.com> References: <4770F7F5-BCBB-47B6-A909-263EEFEADFE4@jerris.com> Message-ID: <1497890891.572062.1014326656.673B64EF@webmail.messagingengine.com> What makes mod_fifo better these days? Some reason I was under the impression mod_callcenter is a better choice. Can you please give some real world examples where mod_fifo excels compared to mod_callcenter, or features that are not possible to implement with the latter? Thanks Mike On Mon, Jun 19, 2017, at 09:34 AM, Michael Jerris wrote: > mod_fifo is a much more feature rich version of a call queue than mod_callcenter. You might want to check that out instead. > From mike at jerris.com Mon Jun 19 18:08:48 2017 From: mike at jerris.com (Michael Jerris) Date: Mon, 19 Jun 2017 14:08:48 -0400 Subject: [Freeswitch-users] When calling out , freeswitch doesn't get DTMF ? In-Reply-To: References: <15c898b5d58.2779.b07ebdf329620b8089087c7205b03f01@xbipin.com> <310cf71c.8fc3.15c8ba9c4f6.Coremail.eastour@163.com> <2a18422f.ef8.15c99b7b9bb.Coremail.eastour@163.com> <6c11dd0f.c2c1.15c9c727dc7.Coremail.eastour@163.com> <2f48505d.d9c5.15ca1d28424.Coremail.eastour@163.com> <59401B92020000310000A34D@mail.tedssupply.com> <59412DB4020000310000A381@mail.tedssupply.com> <7d1e3f12.4750.15ca994d89d.Coremail.eastour@163.com> <7611bb49.72be.15caf10d2a0.Coremail.eastour@163.com> Message-ID: <8CCB1A80-0047-4D23-91BC-AFF7FB70036C@jerris.com> its worth testing both most recent master and 1.6.18. > On Jun 17, 2017, at 4:32 AM, Giovanni Maruzzelli wrote: > > > > On 16 June 2017 at 06:02, chenyzhi > wrote: > Yes ,I can hear all the IVR prompt voices correctly. > > I don't think it's a NAT problem ,because both the x-lite and the freeswitch are in the same LAN. > > The sip trace log is in the attatchment. Thank you. > > PS I tested this on another freeswitch box ,version: > FreeSWITCH Version 1.6.16+git~20170403T142423Z~e6d643b29c~32bit (git e6d643b 2017-04-03 14:24:23Z 32bit) > It can detect dtmf on outgoing calls. Maybe this only happens on FreeSWITCH Version 1.9.0+git~20170518T231917Z~a1fc18aee5~32bit (git a1fc18a 2017-05-18 23:19:17Z 32bit) > > > then use the stable version, and open a jira for this issue citing the master version you are using > -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Mon Jun 19 18:10:07 2017 From: mike at jerris.com (Michael Jerris) Date: Mon, 19 Jun 2017 14:10:07 -0400 Subject: [Freeswitch-users] enable Portal, error 404 In-Reply-To: <8779966.9K6fzknjRZ@stefan-ubu> References: <8779966.9K6fzknjRZ@stefan-ubu> Message-ID: bad credentials? > On Jun 17, 2017, at 10:55 AM, Stefan Fuhrmann wrote: > > Hello all, > > Im new to freeswitch and have to ask, how can I enable the portal? > I installed the debian installation and followed the instruction from wiki to > enable: > https://wiki.freeswitch.org/wiki/Freeswitch_Portal > It is based on mod_xml_rpc, the module is built by default but not loaded, so > you just need to load it (un-comment it in conf/autoload_configs/ > modules.conf.xml) > > load mod_xml_rpc > > When I trying to access > ip:8080/portal/index.html > after login Im getting: > error 404 > > What Im missing? > > Can somone help? > > Tia > Stefan From mike at jerris.com Mon Jun 19 18:10:56 2017 From: mike at jerris.com (Michael Jerris) Date: Mon, 19 Jun 2017 14:10:56 -0400 Subject: [Freeswitch-users] timer not properly configured In-Reply-To: References: <70e56fd8-4c40-53e1-2de2-bd8b879daad2@madovsky.org> Message-ID: FS-10405, fixed in master. Its just cosmetic. > On Jun 17, 2017, at 1:29 PM, Anthony Minessale wrote: > > Jira jira jira > On Sat, Jun 17, 2017 at 7:55 AM Madovsky > wrote: > Hi all, > > last today git gives > > switch_core_timer.c:117 Timer is not properly configured > > everytime a call is hangup. > > show timer gives > > type,name,ikey > timer,soft,CORE_SOFTTIMER_MODULE > > 1 total. > > > Thanks > > F > -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Mon Jun 19 18:19:26 2017 From: mike at jerris.com (Michael Jerris) Date: Mon, 19 Jun 2017 14:19:26 -0400 Subject: [Freeswitch-users] Debian Stretch In-Reply-To: References: <733A7BE7-6E66-4BD8-8736-B6DDBFE695A4@jerris.com> Message-ID: <9BCA607F-C887-4E6C-8A0E-DD4A7305D214@jerris.com> Of course, I’m excited for that too… and as soon as they are ready we will let you know > On Jun 18, 2017, at 2:59 PM, Volodymyr Fedorov wrote: > > Hi Michael, > so from today Stretch is current stable it will be really cool to have packages from freeswitch repository . > > Thanks! > > On Tue, Jun 13, 2017 at 4:14 AM, Michael Jerris > wrote: > announcements will come out when we have real dates. > >> On Jun 12, 2017, at 9:32 PM, Peter Rex > wrote: >> >> Thanks Michael. Hate to do this to you, but is there an estimate on 1.8 timeframe? Mailing list shows people were talking about configs and feature requests in January, but can't see much else. Maybe I'm not looking in the right place. >> >> On Mon, Jun 12, 2017 at 6:44 PM, Michael Jerris > wrote: >> Stretch won’t build yet. I’ll have some patches over the next few weeks to fix that. 1.8 when released will likely target Stretch as its primary but still a bunch of testing to do. The patches to fix build for stretch will go back into 1.6 branch, once they are complete and tested. >> >> >>> On Jun 12, 2017, at 8:28 PM, Peter Rex > wrote: >>> >>> Stretch is the new stable on Saturday. I've looked through Confluence and the mailing lists but I can't find anything relevant. I see interesting possibilities at http://files.freeswitch.org/repo/deb , but I thought I would ask the mailing list if there's a plan yet to add or move the _production_ build to Stretch. > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Mon Jun 19 18:27:08 2017 From: mike at jerris.com (Michael Jerris) Date: Mon, 19 Jun 2017 14:27:08 -0400 Subject: [Freeswitch-users] Callcenter module, can I originate call for an agent in uuid-standby mode ? In-Reply-To: <1497890891.572062.1014326656.673B64EF@webmail.messagingengine.com> References: <4770F7F5-BCBB-47B6-A909-263EEFEADFE4@jerris.com> <1497890891.572062.1014326656.673B64EF@webmail.messagingengine.com> Message-ID: mod_fifo has ALWAYS been superior, people assume otherwise because of the name. Check it out, its pretty powerful. mod_callcenter was written because people had a hard time understanding mod_fifo. It supports agent tracking, some skills routing, inbound and outbound agents, etc. If there is stuff missing we should really sort getting it into mod_fifo and abandon mod_callcenter. > On Jun 19, 2017, at 12:48 PM, Michael Avers wrote: > > What makes mod_fifo better these days? Some reason I was under the impression mod_callcenter is a better choice. Can you please give some real world examples where mod_fifo excels compared to mod_callcenter, or features that are not possible to implement with the latter? > > Thanks > Mike > > On Mon, Jun 19, 2017, at 09:34 AM, Michael Jerris wrote: >> mod_fifo is a much more feature rich version of a call queue than mod_callcenter. You might want to check that out instead. >> From agubbe at gmail.com Tue Jun 20 08:45:42 2017 From: agubbe at gmail.com (=?UTF-8?Q?Agust=C3=AD_Ubalde_Bellot?=) Date: Tue, 20 Jun 2017 10:45:42 +0200 Subject: [Freeswitch-users] Freeswitch sslv3 support In-Reply-To: References: Message-ID: Hi Michael, I have performed several connection tests forcing the sslv3 protocol over secure web sockets and the connection is established. Instead, the same test connecting to the TLS listening port, the connection is not set. The protocol is successfully disabled in the configuration. The version of FreeSWITCH I'm testing is 1.5.14. Is there any way to prove that the sslv3 protocol is actually disabled in this release? Thanks, Agustí 2017-06-15 10:07 GMT+02:00 Agustí Ubalde Bellot : > Hi Brian, > > Is possible to disable for web socket secure connections too? > > > Thanks, > Agustí > > 2017-06-13 13:24 GMT+02:00 Agustí Ubalde Bellot : > >> Hi all, >> >> Is there a FreeSWITCH update where sslv3 support is disabled? >> >> >> Thanks, >> Agustí >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From italo at freeswitch.org Tue Jun 20 12:45:22 2017 From: italo at freeswitch.org (=?UTF-8?B?w410YWxvIFJvc3Np?=) Date: Tue, 20 Jun 2017 09:45:22 -0300 Subject: [Freeswitch-users] Calling on the community for Bug Marshals In-Reply-To: References: Message-ID: This is a great opportunity to learn and to be an expert in FreeSWITCH. This was how I learn a lot! :-) On Mon, Jun 5, 2017 at 4:39 PM, Brian West wrote: > FreeSWITCHers, > > We are in need of a few good bug marshals, We are trying to get 1.8 ready > and out the door and the more help we have testing and working thru patches > on JIRA the quicker it will arrive. If you're interested in helping us out > email me directly. We are also considering bringing back a few days a week > we are sitting in 888 and helping the community out with issues pending in > JIRA. > > Also we are only about 2600 short on the gofund me for the Allison > prompts, which will be delivered sometime this week. ;) So help us get > over that last little bit this week. > > Thanks, > > -- > > *Brian West* > brian at freeswitch.org > > *Twitter: @FreeSWITCH , @briankwest* > > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > Book a phone call (CST) > > Allison prompts for FreeSWITCH: > > *https://www.gofundme.com/allison-prompts-for-freeswitch* > > > Got Bugs? Report them here ! | Reddit: > /r/freeswitch > > *T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002 <(918)%20420-9002> > | *M:*+1918424WEST (9378) > *Skype:*briankwest > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Ítalo Rossi italo at freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From raman.chv at gmail.com Tue Jun 20 13:07:08 2017 From: raman.chv at gmail.com (Ram) Date: Tue, 20 Jun 2017 18:37:08 +0530 Subject: [Freeswitch-users] Record-routes in NOTIFY In-Reply-To: References: Message-ID: Raised the issue in JIRA: https://freeswitch.org/jira/browse/FS-10393 On Tue, Jun 13, 2017 at 7:27 PM, Brian West wrote: > Any bug reports belong on JIRA, https://freeswitch.org/ > confluence/display/FREESWITCH/Reporting+Bugs+to+JIRA > > Thanks, > > > On Tue, Jun 13, 2017 at 3:35 AM, Ram wrote: > >> Hi, >> >> Record routes in SUBSCRIBE is not honored in NOTIFY, In my case i am >> having 3 record routes in SUBSCRIBE, but only one i.e top record route is >> used for NOTIFY is causing routing issue. I am using freeswitch version >> 1.6.17 for testing. >> >> Following is the trace for SUBSCRIBE and NOTIFY at freeswitch. >> >> T 2017/06/13 08:20:14.868294 10.1.30.27:55503 -> 10.2.30.63:5060 [AP] >> SUBSCRIBE sip:500 at 52.64.221.219:5060;transport=tcp SIP/2.0 >> Record-Route: >> Record-Route: >> Record-Route: >> Via: SIP/2.0/TCP 10.1.30.199;branch=z9hG4bKc038 >> .62a9c22ab84694b453503a45210a1392.0;i=1 >> Via: SIP/2.0/TCP 10.1.30.174;branch=z9hG4bKc038 >> .8d466a0ae4f1821a3f0d08e6602cdadc.0;i=82 >> Via: SIP/2.0/TLS 10.1.30.146:51890;received=10. >> 1.30.146;rport=51890;branch=z9hG4bKd1633fda00007 >> From: "RamanTest";tag=46baee66 >> To: "sip:500 at freeconf.com"; >> tag=pv4B8Q9XUDtgD >> Call-ID: 2d118609-1 at 10.1.30.180 >> CSeq: 1805684444 SUBSCRIBE >> Max-Forwards: 69 >> Contact: "RamanTest" >> User-Agent: TestConference >> Event: conference >> Expires: 3600 >> Allow: INVITE,ACK,BYE,CANCEL,REFER,NOTIFY,OPTIONS,PRACK,UPDATE,INFO >> ,MESSAGE,SUBSCRIBE,PUBLISH >> Allow-Events: refer, presence >> Supported: replaces, timer, gruu, join >> Date: Tue, 13 Jun 2017 08:24:00 GMT >> Content-Length: 0 >> >> >> ## >> T 2017/06/13 08:20:14.873026 10.2.30.63:5060 -> 10.1.30.27:55503 [AP] >> SIP/2.0 202 Accepted >> Via: SIP/2.0/TCP 10.1.30.199;branch=z9hG4bKc038 >> .62a9c22ab84694b453503a45210a1392.0;i=1;received=10.1.30.27;rport=55503 >> Via: SIP/2.0/TCP 10.1.30.174;branch=z9hG4bKc038 >> .8d466a0ae4f1821a3f0d08e6602cdadc.0;i=82 >> Via: SIP/2.0/TLS 10.1.30.146:51890;received=10. >> 1.30.146;rport=51890;branch=z9hG4bKd1633fda00007 >> Record-Route: >> Record-Route: >> Record-Route: >> From: "RamanTest" ;tag=46baee66 >> To: "sip:500 at freeconf.com" > t=tls>;tag=pv4B8Q9XUDtgD >> Call-ID: 2d118609-1 at 10.1.30.180 >> CSeq: 1805684444 SUBSCRIBE >> Contact: >> Expires: 3600 >> User-Agent: FreeSWITCH-mod_sofia/1.6.17~64bit >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, path, replaces >> Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, >> line-seize, call-info, sla, include-session-description, presence.winfo, >> message-summary, refer >> Subscription-State: active;expires=3600 >> Content-Length: 0 >> >> >> #### >> T 2017/06/13 08:20:15.173591 10.2.30.63:58879 -> 10.1.30.27:5060 [AP] >> NOTIFY sip:ramantest at 10.1.30.146:51890;transport=tls SIP/2.0 >> Via: SIP/2.0/TCP 52.64.221.219;rport;branch=z9hG4bKvr9Kyp8Fe829g >> Route: ;transport=tcp;ftag=46baee66;lr >> Record-Route: ;transport=tcp;ftag=46baee66;lr >> Max-Forwards: 70 >> From: "sip:500 at freeconf.com" > t=tls>;tag=pv4B8Q9XUDtgD;tag=pv4B8Q9XUDtgD >> To: "RamanTest" ;tag=46baee66 >> Call-ID: 2d118609-1 at 10.1.30.180 >> CSeq: 705660701 NOTIFY >> Contact: ;isfocus >> User-Agent: FreeSWITCH-mod_sofia/1.6.17~64bit >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, >> REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE >> Supported: timer, path, replaces >> Event: conference >> Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, >> line-seize, call-info, sla, include-session-description, presence.winfo, >> message-summary, refer >> Subscription-State: active;expires=3600 >> Content-Type: application/conference-info+xml >> Content-Length: 1028 >> >> >> > entity="sip:500 at freeconf.com"> >> >> FreeSWITCH Conference >> >> >> sip:500 at freeconf.com >> >> >> >> >> 1 >> true >> >> >> >> RamanTest >> >> RamanTest >> connected >> >> 2017-06-13T08:20:13+00:00 >> >> >> audio >> 4048072604 <(404)%20807-2604> >> sendrecv >> >> >> >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > > *Brian West* > brian at freeswitch.org > > *Twitter: @FreeSWITCH , @briankwest* > > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > Book a phone call (CST) > > Allison prompts for FreeSWITCH: > > *https://www.gofundme.com/allison-prompts-for-freeswitch* > > > Got Bugs? Report them here ! | Reddit: > /r/freeswitch > > *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) > *Skype:*briankwest > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From stefan at fuhrmann.homedns.org Mon Jun 19 19:40:44 2017 From: stefan at fuhrmann.homedns.org (Stefan Fuhrmann) Date: Mon, 19 Jun 2017 21:40:44 +0200 Subject: [Freeswitch-users] enable Portal, error 404 In-Reply-To: References: <8779966.9K6fzknjRZ@stefan-ubu> Message-ID: <1794751.l0UOKoQrpU@stefan-ubu> Am Montag, 19. Juni 2017, 14:10:07 CEST schrieb Michael Jerris: > bad credentials? I dont think so: user: freeswitch pass: works that is what I have found in config. Is that wrong? Tia Stefan From juraj.fabo at gmail.com Mon Jun 19 22:14:41 2017 From: juraj.fabo at gmail.com (Juraj Fabo) Date: Tue, 20 Jun 2017 00:14:41 +0200 Subject: [Freeswitch-users] signaling_status Down with libpri Message-ID: Dear list I found this report here and also in JIRA under https://freeswitch.org/jira/browse/OPENZAP-243 where it was closed as fixed after using updated wanpipe. I am facing the very same issue with sangoma card and I am searching for wanpipe version which would provide referenced fix. However, most recent wanpipe I can find on the web is from 2016. Please, would it be possible to provide precise info which module/driver/library needs to be updated and from where the update could be downloaded? Actually I was really trying to followup existing subject from march, but I suspect that this email will create new thread, for that case I would like to apologize in advance. Thank you very much Juraj From nabeel.10.ahmed at gmail.com Tue Jun 20 09:16:13 2017 From: nabeel.10.ahmed at gmail.com (Nabeel Ahmad) Date: Tue, 20 Jun 2017 14:16:13 +0500 Subject: [Freeswitch-users] Call limit on RTP-IP Address Message-ID: Hello all, If we assign more then one RTP-IP parameter to a sip profile , they are used in round robin. Its perfect . I want to know is there a way to use some limit on that ip ? Say i've 5 ip address listening on box , and i want one concurrent call limit on each media ip. How can i set limit on profile level or set from dialplan. I tried to do but it didn't work . Any help or advice will be highly appreciated Thanking all Nabeel. -------------- next part -------------- An HTML attachment was scrubbed... URL: From matt at supportedbusiness.com Tue Jun 20 13:47:42 2017 From: matt at supportedbusiness.com (Matt Broad) Date: Tue, 20 Jun 2017 14:47:42 +0100 Subject: [Freeswitch-users] group_confirm_file multiple files In-Reply-To: References: Message-ID: Thanks Michael! I had looked at mod_file_string before but had overlooked file_string:// I was setting playback_delimiter=! which was not working. For anyone trying to set this using js I have included an example below session.execute("set", "group_confirm_file=file_string://file1.mp3!file2.mp3"); session.execute("set", "group_confirm_key=5"); thanks Matt Matt Broad Tel: +44 (0)203 011 1313 <+44%2020%203011%201313> Web: www.supportedbusiness.com On 19 June 2017 at 17:46, Michael Jerris wrote: > https://freeswitch.org/confluence/display/FREESWITCH/ > mod_dptools%3A+file_string > > > On Jun 15, 2017, at 5:54 AM, Matt Broad > wrote: > > Hi, > > I'm wondering if it is possible to play multiple files using the > group_confirm_file function. > > I have 2 audio files that I would like to play 1 after the other and then > wait for the confirm key. > > > I have tried using mod_file_string, but get an error "Error from mpg123: > File access error. (code 22)", I assume this is due to the fact it is > reading the file string as one file rather than 2 separated by the ! > delimiter. > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From lists at telefaks.de Tue Jun 20 13:53:51 2017 From: lists at telefaks.de (Peter Steinbach) Date: Tue, 20 Jun 2017 15:53:51 +0200 Subject: [Freeswitch-users] Call limit on RTP-IP Address In-Reply-To: References: Message-ID: <594928EF.6080308@telefaks.de> Hello Ahmad, we do it the following way * we have a background job, which periodically (every 1-2 sec) connects via esl to fo Freeswitch and gets it's channel informations. Then we do some calculations based on e.g. call state, domain, profile, ip, other customer information (in our case it's multi-tenant) * for each calculation, we set a memcache key to a specific value (a new call is allowed / not allowed) * then, for a new call, we query the memcache key (mod_memcache) in the dialplan and decide, what to do with this call But - of course - there will also be some other ways to do this, dependent on your specific goal. Best regards Peter On 06/20/17 11:16, Nabeel Ahmad wrote: > Hello all, > If we assign more then one RTP-IP parameter to a sip profile , they > are used in round robin. > Its perfect . I want to know is there a way to use some limit on that ip ? > Say i've 5 ip address listening on box , and i want one concurrent > call limit on each media ip. > How can i set limit on profile level or set from dialplan. > I tried to do but it didn't work . > > Any help or advice will be highly appreciated > Thanking all > Nabeel. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de -------------- next part -------------- An HTML attachment was scrubbed... URL: From krice at freeswitch.org Tue Jun 20 14:05:09 2017 From: krice at freeswitch.org (Ken Rice) Date: Tue, 20 Jun 2017 09:05:09 -0500 Subject: [Freeswitch-users] Call limit on RTP-IP Address In-Reply-To: References: Message-ID: <9cad01d2e9ce$3ab34e50$b019eaf0$@freeswitch.org> Theres no way to limit or select which IP/Port combination is used from the available RTP IP/Port Range in the config you have. The only way to do this would be to create a profile for each IP and then limit the number of calls per profile From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nabeel Ahmad Sent: Tuesday, June 20, 2017 4:16 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Call limit on RTP-IP Address Hello all, If we assign more then one RTP-IP parameter to a sip profile , they are used in round robin. Its perfect . I want to know is there a way to use some limit on that ip ? Say i've 5 ip address listening on box , and i want one concurrent call limit on each media ip. How can i set limit on profile level or set from dialplan. I tried to do but it didn't work . Any help or advice will be highly appreciated Thanking all Nabeel. -------------- next part -------------- An HTML attachment was scrubbed... URL: From khamlichi.khalil at gmail.com Tue Jun 20 14:00:46 2017 From: khamlichi.khalil at gmail.com (Khalil Khamlichi) Date: Tue, 20 Jun 2017 14:00:46 +0000 Subject: [Freeswitch-users] Callcenter module, can I originate call for an agent in uuid-standby mode ? In-Reply-To: References: <4770F7F5-BCBB-47B6-A909-263EEFEADFE4@jerris.com> <1497890891.572062.1014326656.673B64EF@webmail.messagingengine.com> Message-ID: alright, does mod_fifo support some sort of uuid_standby mode ? On Mon, Jun 19, 2017 at 6:27 PM, Michael Jerris wrote: > mod_fifo has ALWAYS been superior, people assume otherwise because of the > name. Check it out, its pretty powerful. mod_callcenter was written > because people had a hard time understanding mod_fifo. It supports agent > tracking, some skills routing, inbound and outbound agents, etc. If there > is stuff missing we should really sort getting it into mod_fifo and abandon > mod_callcenter. > > > > On Jun 19, 2017, at 12:48 PM, Michael Avers > wrote: > > > > What makes mod_fifo better these days? Some reason I was under the > impression mod_callcenter is a better choice. Can you please give some real > world examples where mod_fifo excels compared to mod_callcenter, or > features that are not possible to implement with the latter? > > > > Thanks > > Mike > > > > On Mon, Jun 19, 2017, at 09:34 AM, Michael Jerris wrote: > >> mod_fifo is a much more feature rich version of a call queue than > mod_callcenter. You might want to check that out instead. > >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From nabeel.10.ahmed at gmail.com Tue Jun 20 14:11:53 2017 From: nabeel.10.ahmed at gmail.com (Nabeel Ahmad) Date: Tue, 20 Jun 2017 19:11:53 +0500 Subject: [Freeswitch-users] Call limit on RTP-IP Address In-Reply-To: <594928EF.6080308@telefaks.de> References: <594928EF.6080308@telefaks.de> Message-ID: Hi Peter, So far i also came to something similar. i've all ips in db with status enable/disable. on channel answer event , same ip toggles its status. then updating profile rtp-ip variable and rescan profile. and on destroy event again update status in Db and rescan the profile. If i ignore frequent calls to DB , still i am not getting what will happen if all ips are used . How will i restrict calls more then IPs i've . (can't touch global session limit as there are other profiles where i dont want to limit anything ). So i thought to ask there must be some better way to do it. @Ken : each profile with their own RTP-iP can also work , how to limit number of calls per profile ? On Tue, Jun 20, 2017 at 6:53 PM, Peter Steinbach wrote: > Hello Ahmad, > > we do it the following way > > - we have a background job, which periodically (every 1-2 sec) > connects via esl to fo Freeswitch and gets it's channel informations. Then > we do some calculations based on e.g. call state, domain, profile, ip, > other customer information (in our case it's multi-tenant) > - for each calculation, we set a memcache key to a specific value (a > new call is allowed / not allowed) > - then, for a new call, we query the memcache key (mod_memcache) in > the dialplan and decide, what to do with this call > > But - of course - there will also be some other ways to do this, dependent > on your specific goal. > > Best regards > Peter > > > On 06/20/17 11:16, Nabeel Ahmad wrote: > > Hello all, > If we assign more then one RTP-IP parameter to a sip profile , they are > used in round robin. > Its perfect . I want to know is there a way to use some limit on that ip ? > Say i've 5 ip address listening on box , and i want one concurrent call > limit on each media ip. > How can i set limit on profile level or set from dialplan. > I tried to do but it didn't work . > > Any help or advice will be highly appreciated > Thanking all > Nabeel. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > -- > With kind regards > Peter Steinbach > > Telefaks Services GmbHmailto:lists (att) telefaks.de > Internet: www.telefaks.de > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From me at nevian.org Tue Jun 20 14:17:24 2017 From: me at nevian.org (Serge S. Yuriev) Date: Tue, 20 Jun 2017 17:17:24 +0300 Subject: [Freeswitch-users] Call limit on RTP-IP Address In-Reply-To: <9cad01d2e9ce$3ab34e50$b019eaf0$@freeswitch.org> References: <9cad01d2e9ce$3ab34e50$b019eaf0$@freeswitch.org> Message-ID: <5a52f788-d3b1-a3fb-4e0d-daeb247eb782@nevian.org> Is this multi-IP config expected to work this way or it's eventuality/bug? On 20/06/17 17:05, Ken Rice wrote: > Theres no way to limit or select which IP/Port combination is used from > the available RTP IP/Port Range in the config you have. The only way to > do this would be to create a profile for each IP and then limit the > number of calls per profile > > *From:* FreeSWITCH-users > [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of > *Nabeel Ahmad > *Sent:* Tuesday, June 20, 2017 4:16 AM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* [Freeswitch-users] Call limit on RTP-IP Address > > Hello all, > > If we assign more then one RTP-IP parameter to a sip profile , they are > used in round robin. > > Its perfect . I want to know is there a way to use some limit on that ip ? > > Say i've 5 ip address listening on box , and i want one concurrent call > limit on each media ip. > > How can i set limit on profile level or set from dialplan. > > I tried to do but it didn't work . > > Any help or advice will be highly appreciated > > Thanking all > > Nabeel. -- Serge S. Yuriev Lead VoIP engineer From david.villasmil.work at gmail.com Tue Jun 20 14:30:35 2017 From: david.villasmil.work at gmail.com (David Villasmil) Date: Tue, 20 Jun 2017 14:30:35 +0000 Subject: [Freeswitch-users] Calling on the community for Bug Marshals In-Reply-To: References: Message-ID: Anything i can help with? On Tue, Jun 20, 2017 at 2:46 PM Ítalo Rossi wrote: > This is a great opportunity to learn and to be an expert in FreeSWITCH. > This was how I learn a lot! > > :-) > > On Mon, Jun 5, 2017 at 4:39 PM, Brian West wrote: > >> FreeSWITCHers, >> >> We are in need of a few good bug marshals, We are trying to get 1.8 ready >> and out the door and the more help we have testing and working thru patches >> on JIRA the quicker it will arrive. If you're interested in helping us out >> email me directly. We are also considering bringing back a few days a week >> we are sitting in 888 and helping the community out with issues pending in >> JIRA. >> >> Also we are only about 2600 short on the gofund me for the Allison >> prompts, which will be delivered sometime this week. ;) So help us get >> over that last little bit this week. >> >> Thanks, >> >> -- >> >> *Brian West* >> brian at freeswitch.org >> >> *Twitter: @FreeSWITCH , @briankwest* >> >> http://www.freeswitchbook.com >> http://www.freeswitchcookbook.com >> >> Book a phone call (CST) >> >> Allison prompts for FreeSWITCH: >> >> *https://www.gofundme.com/allison-prompts-for-freeswitch* >> >> >> Got Bugs? Report them here ! | Reddit: >> /r/freeswitch >> >> *T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002 <(918)%20420-9002> >> | *M:*+1918424WEST (9378) >> *Skype:*briankwest >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Ítalo Rossi > italo at freeswitch.org > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Tue Jun 20 14:31:40 2017 From: david.villasmil.work at gmail.com (David Villasmil) Date: Tue, 20 Jun 2017 14:31:40 +0000 Subject: [Freeswitch-users] enable Portal, error 404 In-Reply-To: <1794751.l0UOKoQrpU@stefan-ubu> References: <8779966.9K6fzknjRZ@stefan-ubu> <1794751.l0UOKoQrpU@stefan-ubu> Message-ID: Does netstat shows the port in use? On Tue, Jun 20, 2017 at 3:29 PM Stefan Fuhrmann wrote: > Am Montag, 19. Juni 2017, 14:10:07 CEST schrieb Michael Jerris: > > bad credentials? > > I dont think so: > user: freeswitch > pass: works > > that is what I have found in config. > Is that wrong? > > Tia > Stefan > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From krice at freeswitch.org Tue Jun 20 14:40:49 2017 From: krice at freeswitch.org (Ken Rice) Date: Tue, 20 Jun 2017 09:40:49 -0500 Subject: [Freeswitch-users] Call limit on RTP-IP Address In-Reply-To: <5a52f788-d3b1-a3fb-4e0d-daeb247eb782@nevian.org> References: <9cad01d2e9ce$3ab34e50$b019eaf0$@freeswitch.org> <5a52f788-d3b1-a3fb-4e0d-daeb247eb782@nevian.org> Message-ID: <9d7b01d2e9d3$366f97e0$a34ec7a0$@freeswitch.org> This is not a bug... there is just no way to select which IP or port is used nor is there an effective way to limit it short of just killing the call due to the way the IP/Port allocation works in the RTP stack currently. -----Original Message----- From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Serge S. Yuriev Sent: Tuesday, June 20, 2017 9:17 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Call limit on RTP-IP Address Is this multi-IP config expected to work this way or it's eventuality/bug? On 20/06/17 17:05, Ken Rice wrote: > Theres no way to limit or select which IP/Port combination is used > from the available RTP IP/Port Range in the config you have. The only > way to do this would be to create a profile for each IP and then limit > the number of calls per profile > > *From:* FreeSWITCH-users > [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of > *Nabeel Ahmad > *Sent:* Tuesday, June 20, 2017 4:16 AM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* [Freeswitch-users] Call limit on RTP-IP Address > > Hello all, > > If we assign more then one RTP-IP parameter to a sip profile , they > are used in round robin. > > Its perfect . I want to know is there a way to use some limit on that ip ? > > Say i've 5 ip address listening on box , and i want one concurrent > call limit on each media ip. > > How can i set limit on profile level or set from dialplan. > > I tried to do but it didn't work . > > Any help or advice will be highly appreciated > > Thanking all > > Nabeel. -- Serge S. Yuriev Lead VoIP engineer _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From nabeel.10.ahmed at gmail.com Tue Jun 20 14:20:46 2017 From: nabeel.10.ahmed at gmail.com (Nabeel Ahmad) Date: Tue, 20 Jun 2017 19:20:46 +0500 Subject: [Freeswitch-users] Call limit on RTP-IP Address In-Reply-To: <5a52f788-d3b1-a3fb-4e0d-daeb247eb782@nevian.org> References: <9cad01d2e9ce$3ab34e50$b019eaf0$@freeswitch.org> <5a52f788-d3b1-a3fb-4e0d-daeb247eb782@nevian.org> Message-ID: Its never expected to work this WAY Nor its a Bug. Its a kind of feature request. As FS is a flexible product which allows to do things in many ways , trying to find better approach to do it. On Tue, Jun 20, 2017 at 7:17 PM, Serge S. Yuriev wrote: > Is this multi-IP config expected to work this way or it's eventuality/bug? > > On 20/06/17 17:05, Ken Rice wrote: > >> Theres no way to limit or select which IP/Port combination is used from >> the available RTP IP/Port Range in the config you have. The only way to do >> this would be to create a profile for each IP and then limit the number of >> calls per profile >> >> *From:* FreeSWITCH-users [mailto:freeswitch-users-bounc >> es at lists.freeswitch.org] *On Behalf Of *Nabeel Ahmad >> *Sent:* Tuesday, June 20, 2017 4:16 AM >> *To:* freeswitch-users at lists.freeswitch.org >> *Subject:* [Freeswitch-users] Call limit on RTP-IP Address >> >> Hello all, >> >> If we assign more then one RTP-IP parameter to a sip profile , they are >> used in round robin. >> >> Its perfect . I want to know is there a way to use some limit on that ip ? >> >> Say i've 5 ip address listening on box , and i want one concurrent call >> limit on each media ip. >> >> How can i set limit on profile level or set from dialplan. >> >> I tried to do but it didn't work . >> >> Any help or advice will be highly appreciated >> >> Thanking all >> >> Nabeel. >> > > > -- > Serge S. Yuriev > Lead VoIP engineer > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From krice at freeswitch.org Tue Jun 20 14:42:07 2017 From: krice at freeswitch.org (Ken Rice) Date: Tue, 20 Jun 2017 09:42:07 -0500 Subject: [Freeswitch-users] Call limit on RTP-IP Address In-Reply-To: References: <594928EF.6080308@telefaks.de> Message-ID: <9d7c01d2e9d3$64cc0830$2e641890$@freeswitch.org> https://freeswitch.org/confluence/display/FREESWITCH/mod_dptools%3A+Limit once you have it on a profile you can limit on the profile name using the built in limit framework… this doesn’t work with the multi-rtpip setup as there is no way to indicate which one is to be used before calling bridge From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Nabeel Ahmad Sent: Tuesday, June 20, 2017 9:12 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Call limit on RTP-IP Address Hi Peter, So far i also came to something similar. i've all ips in db with status enable/disable. on channel answer event , same ip toggles its status. then updating profile rtp-ip variable and rescan profile. and on destroy event again update status in Db and rescan the profile. If i ignore frequent calls to DB , still i am not getting what will happen if all ips are used . How will i restrict calls more then IPs i've . (can't touch global session limit as there are other profiles where i dont want to limit anything ). So i thought to ask there must be some better way to do it. @Ken : each profile with their own RTP-iP can also work , how to limit number of calls per profile ? On Tue, Jun 20, 2017 at 6:53 PM, Peter Steinbach > wrote: Hello Ahmad, we do it the following way * we have a background job, which periodically (every 1-2 sec) connects via esl to fo Freeswitch and gets it's channel informations. Then we do some calculations based on e.g. call state, domain, profile, ip, other customer information (in our case it's multi-tenant) * for each calculation, we set a memcache key to a specific value (a new call is allowed / not allowed) * then, for a new call, we query the memcache key (mod_memcache) in the dialplan and decide, what to do with this call But - of course - there will also be some other ways to do this, dependent on your specific goal. Best regards Peter On 06/20/17 11:16, Nabeel Ahmad wrote: Hello all, If we assign more then one RTP-IP parameter to a sip profile , they are used in round robin. Its perfect . I want to know is there a way to use some limit on that ip ? Say i've 5 ip address listening on box , and i want one concurrent call limit on each media ip. How can i set limit on profile level or set from dialplan. I tried to do but it didn't work . Any help or advice will be highly appreciated Thanking all Nabeel. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From achinthau at gmail.com Tue Jun 20 15:05:06 2017 From: achinthau at gmail.com (Achintha) Date: Tue, 20 Jun 2017 20:35:06 +0530 Subject: [Freeswitch-users] switch_core_sqldb.c:2987 invalied cdr data, call not recoverd Message-ID: hi all, I configured two freeswitch servers (freeswitch 1.6.18 on debian 8 ) with call recovery feature. It is working properly on extension to extension , out bound and IVR Calls. Then i tried to land an incoming call, the call gets connected to the queue and then freeswitch generate a call to agent and bridge it with queued call and both sides can hear properly. Then i crashed the primary server, call got landed to the second freeswitch server but rtp is not functioning, and in the second freeswitch console, it printed "switch_core_sqldb.c:2987 invalied cdr data, call not recovered". But the call was not disconnected. I used the following configurations *switch.conf.xml : * switch name is same on both servers core-db-dsn and core-recovery-db-dsn configured with pgsql *both sip profiles:* odbc-dsn configured with pgsql track-calls elabled please provide me a solution to sort out this. -- Best Regards.. Achintha Udukumbura -------------- next part -------------- An HTML attachment was scrubbed... URL: From me at nevian.org Tue Jun 20 15:35:14 2017 From: me at nevian.org (Serge S. Yuriev) Date: Tue, 20 Jun 2017 18:35:14 +0300 Subject: [Freeswitch-users] Call limit on RTP-IP Address In-Reply-To: <9d7b01d2e9d3$366f97e0$a34ec7a0$@freeswitch.org> References: <9cad01d2e9ce$3ab34e50$b019eaf0$@freeswitch.org> <5a52f788-d3b1-a3fb-4e0d-daeb247eb782@nevian.org> <9d7b01d2e9d3$366f97e0$a34ec7a0$@freeswitch.org> Message-ID: I mean if we can define multiple IPs at all. I was under impression parser uses only one last defined. On 20/06/17 17:40, Ken Rice wrote: > This is not a bug... there is just no way to select which IP or port is used nor is there an effective way to limit it short of just killing the call due to the way the IP/Port allocation works in the RTP stack currently. > > -----Original Message----- > From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Serge S. Yuriev > Sent: Tuesday, June 20, 2017 9:17 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Call limit on RTP-IP Address > > Is this multi-IP config expected to work this way or it's eventuality/bug? > > On 20/06/17 17:05, Ken Rice wrote: >> Theres no way to limit or select which IP/Port combination is used >> from the available RTP IP/Port Range in the config you have. The only >> way to do this would be to create a profile for each IP and then limit >> the number of calls per profile >> >> *From:* FreeSWITCH-users >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of >> *Nabeel Ahmad >> *Sent:* Tuesday, June 20, 2017 4:16 AM >> *To:* freeswitch-users at lists.freeswitch.org >> *Subject:* [Freeswitch-users] Call limit on RTP-IP Address >> >> Hello all, >> >> If we assign more then one RTP-IP parameter to a sip profile , they >> are used in round robin. >> >> Its perfect . I want to know is there a way to use some limit on that ip ? >> >> Say i've 5 ip address listening on box , and i want one concurrent >> call limit on each media ip. >> >> How can i set limit on profile level or set from dialplan. >> >> I tried to do but it didn't work . >> >> Any help or advice will be highly appreciated >> >> Thanking all >> >> Nabeel. > > > -- > Serge S. Yuriev > Lead VoIP engineer > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Serge S. Yuriev Lead VoIP engineer From mike at jerris.com Tue Jun 20 15:41:23 2017 From: mike at jerris.com (Michael Jerris) Date: Tue, 20 Jun 2017 11:41:23 -0400 Subject: [Freeswitch-users] Freeswitch sslv3 support In-Reply-To: References: Message-ID: Wait… you are testing against some ancient dev version and not current release? Is that a typo? If not, this makes no sense at all, please explain. > On Jun 20, 2017, at 4:45 AM, Agustí Ubalde Bellot wrote: > > Hi Michael, > > I have performed several connection tests forcing the sslv3 protocol over secure web sockets and the connection is established. Instead, the same test connecting to the TLS listening port, the connection is not set. The protocol is successfully disabled in the configuration. > The version of FreeSWITCH I'm testing is 1.5.14. Is there any way to prove that the sslv3 protocol is actually disabled in this release? > > > Thanks, > Agustí > > 2017-06-15 10:07 GMT+02:00 Agustí Ubalde Bellot >: > Hi Brian, > > Is possible to disable for web socket secure connections too? > > > Thanks, > Agustí > > 2017-06-13 13:24 GMT+02:00 Agustí Ubalde Bellot >: > Hi all, > > Is there a FreeSWITCH update where sslv3 support is disabled? > > > Thanks, > Agustí > -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Tue Jun 20 15:42:38 2017 From: mike at jerris.com (Michael Jerris) Date: Tue, 20 Jun 2017 11:42:38 -0400 Subject: [Freeswitch-users] signaling_status Down with libpri In-Reply-To: References: Message-ID: <59FD8240-CD8B-48FC-A165-819FBE5C9CA8@jerris.com> If you are looking for help setting up sangoma software, please contact sangoma support. They provide their own support for their hardware and drivers. > On Jun 19, 2017, at 6:14 PM, Juraj Fabo wrote: > > Dear list > > > I found this report here and also in JIRA under > https://freeswitch.org/jira/browse/OPENZAP-243 where it was closed as > fixed after using updated wanpipe. > > I am facing the very same issue with sangoma card and I am searching > for wanpipe version which would provide referenced fix. > > However, most recent wanpipe I can find on the web is from 2016. > > Please, would it be possible to provide precise info which > module/driver/library needs to be updated and from where the update > could be downloaded? > > > Actually I was really trying to followup existing subject from march, > but I suspect that this email will create new thread, for that case I > would like to apologize in advance. > > Thank you very much From mike at jerris.com Tue Jun 20 15:43:29 2017 From: mike at jerris.com (Michael Jerris) Date: Tue, 20 Jun 2017 11:43:29 -0400 Subject: [Freeswitch-users] Callcenter module, can I originate call for an agent in uuid-standby mode ? In-Reply-To: References: <4770F7F5-BCBB-47B6-A909-263EEFEADFE4@jerris.com> <1497890891.572062.1014326656.673B64EF@webmail.messagingengine.com> Message-ID: <8C4372DB-F2EB-4614-82EC-185B5AA33C65@jerris.com> Not sure what that means exactly. > On Jun 20, 2017, at 10:00 AM, Khalil Khamlichi wrote: > > alright, does mod_fifo support some sort of uuid_standby mode ? > > On Mon, Jun 19, 2017 at 6:27 PM, Michael Jerris > wrote: > mod_fifo has ALWAYS been superior, people assume otherwise because of the name. Check it out, its pretty powerful. mod_callcenter was written because people had a hard time understanding mod_fifo. It supports agent tracking, some skills routing, inbound and outbound agents, etc. If there is stuff missing we should really sort getting it into mod_fifo and abandon mod_callcenter. > > > > On Jun 19, 2017, at 12:48 PM, Michael Avers > wrote: > > > > What makes mod_fifo better these days? Some reason I was under the impression mod_callcenter is a better choice. Can you please give some real world examples where mod_fifo excels compared to mod_callcenter, or features that are not possible to implement with the latter? > > > > Thanks > > Mike > > > > On Mon, Jun 19, 2017, at 09:34 AM, Michael Jerris wrote: > >> mod_fifo is a much more feature rich version of a call queue than mod_callcenter. You might want to check that out instead. > >> -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Tue Jun 20 15:44:59 2017 From: mike at jerris.com (Michael Jerris) Date: Tue, 20 Jun 2017 11:44:59 -0400 Subject: [Freeswitch-users] switch_core_sqldb.c:2987 invalied cdr data, call not recoverd In-Reply-To: References: Message-ID: <5A5DBC29-DCE6-417F-8B48-2BE9A3390523@jerris.com> would need a bug report on this one with full logs and config and how to reproduce to look into it. > On Jun 20, 2017, at 11:05 AM, Achintha wrote: > > hi all, > > I configured two freeswitch servers (freeswitch 1.6.18 on debian 8 ) with call recovery feature. It is working properly on extension to extension , out bound and IVR Calls. > Then i tried to land an incoming call, the call gets connected to the queue and then freeswitch generate a call to agent and bridge it with queued call and both sides can hear properly. > Then i crashed the primary server, call got landed to the second freeswitch server but rtp is not functioning, and in the second freeswitch console, it printed "switch_core_sqldb.c:2987 invalied cdr data, call not recovered". But the call was not disconnected. > > I used the following configurations > > switch.conf.xml : > > switch name is same on both servers > core-db-dsn and core-recovery-db-dsn configured with pgsql > both sip profiles: > odbc-dsn configured with pgsql > track-calls elabled > > please provide me a solution to sort out this. > > -- > Best Regards.. > Achintha Udukumbura > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From krice at freeswitch.org Tue Jun 20 15:52:29 2017 From: krice at freeswitch.org (Ken Rice) Date: Tue, 20 Jun 2017 10:52:29 -0500 Subject: [Freeswitch-users] Call limit on RTP-IP Address In-Reply-To: References: <9cad01d2e9ce$3ab34e50$b019eaf0$@freeswitch.org> <5a52f788-d3b1-a3fb-4e0d-daeb247eb782@nevian.org> <9d7b01d2e9d3$366f97e0$a34ec7a0$@freeswitch.org> Message-ID: <9e2901d2e9dd$391c0eb0$ab542c10$@freeswitch.org> No Defining multiple RTP IPs has been there for a while.... you don’t define them in different lines, the parser will filter out previous ones, you define them all together The stack will then round robin them. This feature was added several years ago so that FreeSWITCH can handle the required RTP load in traffic flows that can exceed that of 1Gig-E network connections while only using 1 SIP Profile for traffic. It's still useful although less of a requirement with 10GigE coming down in price. (you can now find managed 48port 10GE switches, NICs and Cables on the secondary market for a combined cost under $200/port now. -----Original Message----- From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Serge S. Yuriev Sent: Tuesday, June 20, 2017 10:35 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Call limit on RTP-IP Address I mean if we can define multiple IPs at all. I was under impression parser uses only one last defined. On 20/06/17 17:40, Ken Rice wrote: > This is not a bug... there is just no way to select which IP or port is used nor is there an effective way to limit it short of just killing the call due to the way the IP/Port allocation works in the RTP stack currently. > > -----Original Message----- > From: FreeSWITCH-users > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Serge S. Yuriev > Sent: Tuesday, June 20, 2017 9:17 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Call limit on RTP-IP Address > > Is this multi-IP config expected to work this way or it's eventuality/bug? > > On 20/06/17 17:05, Ken Rice wrote: >> Theres no way to limit or select which IP/Port combination is used >> from the available RTP IP/Port Range in the config you have. The >> only way to do this would be to create a profile for each IP and then >> limit the number of calls per profile >> >> *From:* FreeSWITCH-users >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of >> *Nabeel Ahmad >> *Sent:* Tuesday, June 20, 2017 4:16 AM >> *To:* freeswitch-users at lists.freeswitch.org >> *Subject:* [Freeswitch-users] Call limit on RTP-IP Address >> >> Hello all, >> >> If we assign more then one RTP-IP parameter to a sip profile , they >> are used in round robin. >> >> Its perfect . I want to know is there a way to use some limit on that ip ? >> >> Say i've 5 ip address listening on box , and i want one concurrent >> call limit on each media ip. >> >> How can i set limit on profile level or set from dialplan. >> >> I tried to do but it didn't work . >> >> Any help or advice will be highly appreciated >> >> Thanking all >> >> Nabeel. > > > -- > Serge S. Yuriev > Lead VoIP engineer > > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org > > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > rs > http://www.freeswitch.org > -- Serge S. Yuriev Lead VoIP engineer _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From agubbe at gmail.com Tue Jun 20 15:55:58 2017 From: agubbe at gmail.com (=?UTF-8?Q?Agust=C3=AD_Ubalde_Bellot?=) Date: Tue, 20 Jun 2017 17:55:58 +0200 Subject: [Freeswitch-users] Freeswitch sslv3 support In-Reply-To: References: Message-ID: Hi Michael, Yes, the version I am using is a development version (1.5.14). In any case, I have performed the same tests in version 1.6 and have the same behavior. Instead, the verto module does block the sslv3 protocol. Thanks, Agustí 2017-06-20 10:45 GMT+02:00 Agustí Ubalde Bellot : > Hi Michael, > > I have performed several connection tests forcing the sslv3 protocol over > secure web sockets and the connection is established. Instead, the same > test connecting to the TLS listening port, the connection is not set. The > protocol is successfully disabled in the configuration. > The version of FreeSWITCH I'm testing is 1.5.14. Is there any way to prove > that the sslv3 protocol is actually disabled in this release? > > > Thanks, > Agustí > > 2017-06-15 10:07 GMT+02:00 Agustí Ubalde Bellot : > >> Hi Brian, >> >> Is possible to disable for web socket secure connections too? >> >> >> Thanks, >> Agustí >> >> 2017-06-13 13:24 GMT+02:00 Agustí Ubalde Bellot : >> >>> Hi all, >>> >>> Is there a FreeSWITCH update where sslv3 support is disabled? >>> >>> >>> Thanks, >>> Agustí >>> >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: From joelists at tm.net.uk Tue Jun 20 16:21:52 2017 From: joelists at tm.net.uk (Joseph Waite) Date: Tue, 20 Jun 2017 17:21:52 +0100 Subject: [Freeswitch-users] Radius auth based on IP without sending 407 Proxy authentication required Message-ID: Hi Guys I am trying to configure a Sofia Profile that will not send a 407 Proxy Authentication Required, but will still authenticate the incoming invite via Radius based on the IP address of the INVITE. If I change the Auth_calls to false on the Sofia profile, it doesn’t send the 407, but then it doesn’t authenticate the call. Any help would be most appreciated. Regards From mike at jerris.com Tue Jun 20 16:29:38 2017 From: mike at jerris.com (Michael Jerris) Date: Tue, 20 Jun 2017 12:29:38 -0400 Subject: [Freeswitch-users] Freeswitch sslv3 support In-Reply-To: References: Message-ID: I just reviewed the code. Looks like we disable it all on verto, but not on sofia. should get me a Jira on this (I was just told one just got made)… we should fix that. That being said, the sofia web socket support was basically a demo to prove we could do it before we finished verto, there is little reason to use sip over websockets and I never recommend it. Also, using years old development code should be considered a massive security vulnerability and I would STRONGLY recommend against it. > On Jun 20, 2017, at 11:55 AM, Agustí Ubalde Bellot wrote: > > Hi Michael, > > Yes, the version I am using is a development version (1.5.14). In any case, I have performed the same tests in version 1.6 and have the same behavior. > Instead, the verto module does block the sslv3 protocol. > > > Thanks, > Agustí > > 2017-06-20 10:45 GMT+02:00 Agustí Ubalde Bellot >: > Hi Michael, > > I have performed several connection tests forcing the sslv3 protocol over secure web sockets and the connection is established. Instead, the same test connecting to the TLS listening port, the connection is not set. The protocol is successfully disabled in the configuration. > The version of FreeSWITCH I'm testing is 1.5.14. Is there any way to prove that the sslv3 protocol is actually disabled in this release? > > > Thanks, > Agustí > > 2017-06-15 10:07 GMT+02:00 Agustí Ubalde Bellot >: > Hi Brian, > > Is possible to disable for web socket secure connections too? > > > Thanks, > Agustí > > 2017-06-13 13:24 GMT+02:00 Agustí Ubalde Bellot >: > Hi all, > > Is there a FreeSWITCH update where sslv3 support is disabled? -------------- next part -------------- An HTML attachment was scrubbed... URL: From s.safarov at gmail.com Tue Jun 20 16:32:15 2017 From: s.safarov at gmail.com (Sergey Safarov) Date: Tue, 20 Jun 2017 16:32:15 +0000 Subject: [Freeswitch-users] Radius auth based on IP without sending 407 Proxy authentication required In-Reply-To: References: Message-ID: You can try mod_xml_radius and generate directory record with cidr adribute. вт, 20 июня 2017 г., 19:25 Joseph Waite : > Hi Guys > > I am trying to configure a Sofia Profile that will not send a 407 Proxy > Authentication Required, but will still authenticate the incoming invite > via Radius based on the IP address of the INVITE. > > If I change the Auth_calls to false on the Sofia profile, it doesn’t send > the 407, but then it doesn’t authenticate the call. > > Any help would be most appreciated. > > Regards > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From acheraime at gmail.com Tue Jun 20 16:35:22 2017 From: acheraime at gmail.com (acheraime at gmail.com) Date: Tue, 20 Jun 2017 12:35:22 -0400 Subject: [Freeswitch-users] Calling on the community for Bug Marshals In-Reply-To: References: Message-ID: I would be happy to help. Let me know what the "requirements" are. Sent from my iPhone > On Jun 20, 2017, at 10:30 AM, David Villasmil wrote: > > Anything i can help with? >> On Tue, Jun 20, 2017 at 2:46 PM Ítalo Rossi wrote: >> This is a great opportunity to learn and to be an expert in FreeSWITCH. This was how I learn a lot! >> >> :-) >> >>> On Mon, Jun 5, 2017 at 4:39 PM, Brian West wrote: >>> FreeSWITCHers, >>> >>> We are in need of a few good bug marshals, We are trying to get 1.8 ready and out the door and the more help we have testing and working thru patches on JIRA the quicker it will arrive. If you're interested in helping us out email me directly. We are also considering bringing back a few days a week we are sitting in 888 and helping the community out with issues pending in JIRA. >>> >>> Also we are only about 2600 short on the gofund me for the Allison prompts, which will be delivered sometime this week. ;) So help us get over that last little bit this week. >>> >>> Thanks, >>> >>> -- >>> Brian West >>> brian at freeswitch.org >>> >>> Twitter: @FreeSWITCH , @briankwest >>> >>> http://www.freeswitchbook.com >>> http://www.freeswitchcookbook.com >>> >>> Book a phone call (CST) >>> >>> Allison prompts for FreeSWITCH: >>> >>> https://www.gofundme.com/allison-prompts-for-freeswitch >>> >>> Got Bugs? Report them here! | Reddit: /r/freeswitch >>> >>> T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378) >>> Skype:briankwest >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at 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 at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> >> -- >> Ítalo Rossi >> italo at freeswitch.org >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From s.safarov at gmail.com Tue Jun 20 16:36:22 2017 From: s.safarov at gmail.com (Sergey Safarov) Date: Tue, 20 Jun 2017 16:36:22 +0000 Subject: [Freeswitch-users] Radius auth based on IP without sending 407 Proxy authentication required References: Message-ID: Also required to generate "domain" acl when freeswith starts, executed command "reloadacl" and updated client ip on radius server side вт, 20 июня 2017 г., 19:32 Sergey Safarov : > You can try mod_xml_radius and generate directory record with cidr > adribute. > > вт, 20 июня 2017 г., 19:25 Joseph Waite : > >> Hi Guys >> >> I am trying to configure a Sofia Profile that will not send a 407 Proxy >> Authentication Required, but will still authenticate the incoming invite >> via Radius based on the IP address of the INVITE. >> >> If I change the Auth_calls to false on the Sofia profile, it doesn’t send >> the 407, but then it doesn’t authenticate the call. >> >> Any help would be most appreciated. >> >> Regards >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From joelists at tm.net.uk Tue Jun 20 16:41:01 2017 From: joelists at tm.net.uk (Joseph Waite) Date: Tue, 20 Jun 2017 17:41:01 +0100 Subject: [Freeswitch-users] Radius auth based on IP without sending 407 Proxy authentication required In-Reply-To: References: Message-ID: I am using mod_xml_radius, however my issue is that if I enable auth_calls in profile it sends a 407 Proxy Authentication Required sip message, and if I set auth_calls to false it doesn’t authenticate with Radius, it simply passes call straight into the dial plan. > On 20 Jun 2017, at 17:32, Sergey Safarov wrote: > > You can try mod_xml_radius and generate directory record with cidr adribute. > > > вт, 20 июня 2017 г., 19:25 Joseph Waite >: > Hi Guys > > I am trying to configure a Sofia Profile that will not send a 407 Proxy Authentication Required, but will still authenticate the incoming invite via Radius based on the IP address of the INVITE. > > If I change the Auth_calls to false on the Sofia profile, it doesn’t send the 407, but then it doesn’t authenticate the call. > > Any help would be most appreciated. > > Regards > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From s.safarov at gmail.com Tue Jun 20 17:15:16 2017 From: s.safarov at gmail.com (Sergey Safarov) Date: Tue, 20 Jun 2017 17:15:16 +0000 Subject: [Freeswitch-users] Radius auth based on IP without sending 407 Proxy authentication required In-Reply-To: References: Message-ID: When you generate manually ACL with trusted IP or via mod_xml_radius then you can accept call without 407 message. Also if you add cird attribute to user directory then FreeSwitch can map call to user via cidr attribute. вт, 20 июн. 2017 г. в 19:43, Joseph Waite : > I am using mod_xml_radius, however my issue is that if I enable auth_calls > in profile it sends a 407 Proxy Authentication Required sip message, and if > I set auth_calls to false it doesn’t authenticate with Radius, it simply > passes call straight into the dial plan. > > > On 20 Jun 2017, at 17:32, Sergey Safarov wrote: > > You can try mod_xml_radius and generate directory record with cidr > adribute. > > вт, 20 июня 2017 г., 19:25 Joseph Waite : > >> Hi Guys >> >> I am trying to configure a Sofia Profile that will not send a 407 Proxy >> Authentication Required, but will still authenticate the incoming invite >> via Radius based on the IP address of the INVITE. >> >> If I change the Auth_calls to false on the Sofia profile, it doesn’t send >> the 407, but then it doesn’t authenticate the call. >> >> Any help would be most appreciated. >> >> Regards >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From joelists at tm.net.uk Tue Jun 20 17:22:35 2017 From: joelists at tm.net.uk (Jospeh Waite) Date: Tue, 20 Jun 2017 18:22:35 +0100 Subject: [Freeswitch-users] Radius auth based on IP without sending 407 Proxy authentication required In-Reply-To: References: Message-ID: <84019A4D-1659-496F-8542-BAE10F901641@tm.net.uk> I don’t want to allow certain users, I want all calls that come in on this Sofia profile to not send the 407 but still authenticate via Radius based on the IP. > On 20 Jun 2017, at 18:15, Sergey Safarov wrote: > > When you generate manually ACL with trusted IP or via mod_xml_radius then you can accept call without 407 message. > Also if you add cird attribute to user directory then FreeSwitch can map call to user via cidr attribute. > > вт, 20 июн. 2017 г. в 19:43, Joseph Waite >: > I am using mod_xml_radius, however my issue is that if I enable auth_calls in profile it sends a 407 Proxy Authentication Required sip message, and if I set auth_calls to false it doesn’t authenticate with Radius, it simply passes call straight into the dial plan. > > >> On 20 Jun 2017, at 17:32, Sergey Safarov > wrote: >> >> You can try mod_xml_radius and generate directory record with cidr adribute. >> >> >> вт, 20 июня 2017 г., 19:25 Joseph Waite >: >> Hi Guys >> >> I am trying to configure a Sofia Profile that will not send a 407 Proxy Authentication Required, but will still authenticate the incoming invite via Radius based on the IP address of the INVITE. >> >> If I change the Auth_calls to false on the Sofia profile, it doesn’t send the 407, but then it doesn’t authenticate the call. >> >> Any help would be most appreciated. >> >> Regards >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at 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 at 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 at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From s.safarov at gmail.com Tue Jun 20 18:12:22 2017 From: s.safarov at gmail.com (Sergey Safarov) Date: Tue, 20 Jun 2017 18:12:22 +0000 Subject: [Freeswitch-users] Radius auth based on IP without sending 407 Proxy authentication required In-Reply-To: <84019A4D-1659-496F-8542-BAE10F901641@tm.net.uk> References: <84019A4D-1659-496F-8542-BAE10F901641@tm.net.uk> Message-ID: you can add cidr attribute for all users вт, 20 июн. 2017 г. в 20:25, Jospeh Waite : > I don’t want to allow certain users, I want all calls that come in on this > Sofia profile to not send the 407 but still authenticate via Radius based > on the IP. > > > On 20 Jun 2017, at 18:15, Sergey Safarov wrote: > > When you generate manually ACL with trusted IP or via mod_xml_radius then > you can accept call without 407 message. > Also if you add cird attribute to user directory then FreeSwitch can map > call to user via cidr attribute. > > вт, 20 июн. 2017 г. в 19:43, Joseph Waite : > >> I am using mod_xml_radius, however my issue is that if I enable >> auth_calls in profile it sends a 407 Proxy Authentication Required sip >> message, and if I set auth_calls to false it doesn’t authenticate with >> Radius, it simply passes call straight into the dial plan. >> >> >> On 20 Jun 2017, at 17:32, Sergey Safarov wrote: >> >> You can try mod_xml_radius and generate directory record with cidr >> adribute. >> >> вт, 20 июня 2017 г., 19:25 Joseph Waite : >> >>> Hi Guys >>> >>> I am trying to configure a Sofia Profile that will not send a 407 Proxy >>> Authentication Required, but will still authenticate the incoming invite >>> via Radius based on the IP address of the INVITE. >>> >>> If I change the Auth_calls to false on the Sofia profile, it doesn’t >>> send the 407, but then it doesn’t authenticate the call. >>> >>> Any help would be most appreciated. >>> >>> Regards >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From joelists at tm.net.uk Tue Jun 20 18:22:20 2017 From: joelists at tm.net.uk (Joseph Waite) Date: Tue, 20 Jun 2017 19:22:20 +0100 Subject: [Freeswitch-users] Radius auth based on IP without sending 407 Proxy authentication required In-Reply-To: References: <84019A4D-1659-496F-8542-BAE10F901641@tm.net.uk> Message-ID: <7FCEE5B4-6B43-4623-ACC1-1709FFBC32A2@tm.net.uk> There are no users configured on FreeSwitch, all user config is done on JeraSoft VCS and FreeSwitch authenticates via radius! FreeSwitch should allow an INVITE from any IP, not send 407, but authenticate via Radius > On 20 Jun 2017, at 19:12, Sergey Safarov wrote: > > you can add cidr attribute for all users > > вт, 20 июн. 2017 г. в 20:25, Jospeh Waite >: > I don’t want to allow certain users, I want all calls that come in on this Sofia profile to not send the 407 but still authenticate via Radius based on the IP. > > >> On 20 Jun 2017, at 18:15, Sergey Safarov > wrote: >> >> When you generate manually ACL with trusted IP or via mod_xml_radius then you can accept call without 407 message. >> Also if you add cird attribute to user directory then FreeSwitch can map call to user via cidr attribute. >> >> вт, 20 июн. 2017 г. в 19:43, Joseph Waite >: >> I am using mod_xml_radius, however my issue is that if I enable auth_calls in profile it sends a 407 Proxy Authentication Required sip message, and if I set auth_calls to false it doesn’t authenticate with Radius, it simply passes call straight into the dial plan. >> >> >>> On 20 Jun 2017, at 17:32, Sergey Safarov > wrote: >>> >>> You can try mod_xml_radius and generate directory record with cidr adribute. >>> >>> >>> вт, 20 июня 2017 г., 19:25 Joseph Waite >: >>> Hi Guys >>> >>> I am trying to configure a Sofia Profile that will not send a 407 Proxy Authentication Required, but will still authenticate the incoming invite via Radius based on the IP address of the INVITE. >>> >>> If I change the Auth_calls to false on the Sofia profile, it doesn’t send the 407, but then it doesn’t authenticate the call. >>> >>> Any help would be most appreciated. >>> >>> Regards >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From tculjaga at gmail.com Tue Jun 20 18:24:43 2017 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Tue, 20 Jun 2017 20:24:43 +0200 Subject: [Freeswitch-users] Radius auth based on IP without sending 407 Proxy authentication required In-Reply-To: <7FCEE5B4-6B43-4623-ACC1-1709FFBC32A2@tm.net.uk> References: <84019A4D-1659-496F-8542-BAE10F901641@tm.net.uk> <7FCEE5B4-6B43-4623-ACC1-1709FFBC32A2@tm.net.uk> Message-ID: mod_rad_auth if you want just to authenticate. its an application that you can trigger in dialplan regradless if its an incoming or outgoing call .. On 20 June 2017 at 20:22, Joseph Waite wrote: > There are no users configured on FreeSwitch, all user config is done on > JeraSoft VCS and FreeSwitch authenticates via radius! > > FreeSwitch should allow an INVITE from any IP, not send 407, but > authenticate via Radius > > On 20 Jun 2017, at 19:12, Sergey Safarov wrote: > > you can add cidr attribute for all users > > вт, 20 июн. 2017 г. в 20:25, Jospeh Waite : > >> I don’t want to allow certain users, I want all calls that come in on >> this Sofia profile to not send the 407 but still authenticate via Radius >> based on the IP. >> >> >> On 20 Jun 2017, at 18:15, Sergey Safarov wrote: >> >> When you generate manually ACL with trusted IP or via mod_xml_radius then >> you can accept call without 407 message. >> Also if you add cird attribute to user directory then FreeSwitch can map >> call to user via cidr attribute. >> >> вт, 20 июн. 2017 г. в 19:43, Joseph Waite : >> >>> I am using mod_xml_radius, however my issue is that if I enable >>> auth_calls in profile it sends a 407 Proxy Authentication Required sip >>> message, and if I set auth_calls to false it doesn’t authenticate with >>> Radius, it simply passes call straight into the dial plan. >>> >>> >>> On 20 Jun 2017, at 17:32, Sergey Safarov wrote: >>> >>> You can try mod_xml_radius and generate directory record with cidr >>> adribute. >>> >>> вт, 20 июня 2017 г., 19:25 Joseph Waite : >>> >>>> Hi Guys >>>> >>>> I am trying to configure a Sofia Profile that will not send a 407 Proxy >>>> Authentication Required, but will still authenticate the incoming invite >>>> via Radius based on the IP address of the INVITE. >>>> >>>> If I change the Auth_calls to false on the Sofia profile, it doesn’t >>>> send the 407, but then it doesn’t authenticate the call. >>>> >>>> Any help would be most appreciated. >>>> >>>> Regards >>>> ____________________________________________________________ >>>> _____________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From tculjaga at gmail.com Tue Jun 20 18:28:14 2017 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Tue, 20 Jun 2017 20:28:14 +0200 Subject: [Freeswitch-users] Radius auth based on IP without sending 407 Proxy authentication required In-Reply-To: References: <84019A4D-1659-496F-8542-BAE10F901641@tm.net.uk> <7FCEE5B4-6B43-4623-ACC1-1709FFBC32A2@tm.net.uk> Message-ID: i build that module few years ago and it still works as a charm :P have fun, Tihomir. On 20 June 2017 at 20:24, Tihomir Culjaga wrote: > mod_rad_auth if you want just to authenticate. > > its an application that you can trigger in dialplan regradless if its an > incoming or outgoing call .. > > On 20 June 2017 at 20:22, Joseph Waite wrote: > >> There are no users configured on FreeSwitch, all user config is done on >> JeraSoft VCS and FreeSwitch authenticates via radius! >> >> FreeSwitch should allow an INVITE from any IP, not send 407, but >> authenticate via Radius >> >> On 20 Jun 2017, at 19:12, Sergey Safarov wrote: >> >> you can add cidr attribute for all users >> >> вт, 20 июн. 2017 г. в 20:25, Jospeh Waite : >> >>> I don’t want to allow certain users, I want all calls that come in on >>> this Sofia profile to not send the 407 but still authenticate via Radius >>> based on the IP. >>> >>> >>> On 20 Jun 2017, at 18:15, Sergey Safarov wrote: >>> >>> When you generate manually ACL with trusted IP or via mod_xml_radius >>> then you can accept call without 407 message. >>> Also if you add cird attribute to user directory then FreeSwitch can map >>> call to user via cidr attribute. >>> >>> вт, 20 июн. 2017 г. в 19:43, Joseph Waite : >>> >>>> I am using mod_xml_radius, however my issue is that if I enable >>>> auth_calls in profile it sends a 407 Proxy Authentication Required sip >>>> message, and if I set auth_calls to false it doesn’t authenticate with >>>> Radius, it simply passes call straight into the dial plan. >>>> >>>> >>>> On 20 Jun 2017, at 17:32, Sergey Safarov wrote: >>>> >>>> You can try mod_xml_radius and generate directory record with cidr >>>> adribute. >>>> >>>> вт, 20 июня 2017 г., 19:25 Joseph Waite : >>>> >>>>> Hi Guys >>>>> >>>>> I am trying to configure a Sofia Profile that will not send a 407 >>>>> Proxy Authentication Required, but will still authenticate the incoming >>>>> invite via Radius based on the IP address of the INVITE. >>>>> >>>>> If I change the Auth_calls to false on the Sofia profile, it doesn’t >>>>> send the 407, but then it doesn’t authenticate the call. >>>>> >>>>> Any help would be most appreciated. >>>>> >>>>> Regards >>>>> ____________________________________________________________ >>>>> _____________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Tue Jun 20 18:28:46 2017 From: mike at jerris.com (Michael Jerris) Date: Tue, 20 Jun 2017 14:28:46 -0400 Subject: [Freeswitch-users] Radius auth based on IP without sending 407 Proxy authentication required In-Reply-To: <7FCEE5B4-6B43-4623-ACC1-1709FFBC32A2@tm.net.uk> References: <84019A4D-1659-496F-8542-BAE10F901641@tm.net.uk> <7FCEE5B4-6B43-4623-ACC1-1709FFBC32A2@tm.net.uk> Message-ID: the user directory lookup for auth is not done until after we send the 407. To do this without an acl you need to not do auth on the sip profile, and handle it in dial plan instead. > On Jun 20, 2017, at 2:22 PM, Joseph Waite wrote: > > There are no users configured on FreeSwitch, all user config is done on JeraSoft VCS and FreeSwitch authenticates via radius! > > FreeSwitch should allow an INVITE from any IP, not send 407, but authenticate via Radius >> On 20 Jun 2017, at 19:12, Sergey Safarov > wrote: >> >> you can add cidr attribute for all users >> >> вт, 20 июн. 2017 г. в 20:25, Jospeh Waite >: >> I don’t want to allow certain users, I want all calls that come in on this Sofia profile to not send the 407 but still authenticate via Radius based on the IP. >> >> >>> On 20 Jun 2017, at 18:15, Sergey Safarov > wrote: >>> >>> When you generate manually ACL with trusted IP or via mod_xml_radius then you can accept call without 407 message. >>> Also if you add cird attribute to user directory then FreeSwitch can map call to user via cidr attribute. >>> >>> вт, 20 июн. 2017 г. в 19:43, Joseph Waite >: >>> I am using mod_xml_radius, however my issue is that if I enable auth_calls in profile it sends a 407 Proxy Authentication Required sip message, and if I set auth_calls to false it doesn’t authenticate with Radius, it simply passes call straight into the dial plan. >>> >>> >>>> On 20 Jun 2017, at 17:32, Sergey Safarov > wrote: >>>> >>>> You can try mod_xml_radius and generate directory record with cidr adribute. >>>> >>>> >>>> вт, 20 июня 2017 г., 19:25 Joseph Waite >: >>>> Hi Guys >>>> >>>> I am trying to configure a Sofia Profile that will not send a 407 Proxy Authentication Required, but will still authenticate the incoming invite via Radius based on the IP address of the INVITE. >>>> >>>> If I change the Auth_calls to false on the Sofia profile, it doesn’t send the 407, but then it doesn’t authenticate the call. >>>> >>>> Any help would be most appreciated. -------------- next part -------------- An HTML attachment was scrubbed... URL: From joelists at tm.net.uk Tue Jun 20 18:44:09 2017 From: joelists at tm.net.uk (Joseph Waite) Date: Tue, 20 Jun 2017 19:44:09 +0100 Subject: [Freeswitch-users] Radius auth based on IP without sending 407 Proxy authentication required In-Reply-To: References: <84019A4D-1659-496F-8542-BAE10F901641@tm.net.uk> <7FCEE5B4-6B43-4623-ACC1-1709FFBC32A2@tm.net.uk> Message-ID: <8E616BE5-F82E-4FD6-8D98-1E8DC2690F5B@tm.net.uk> Now that sounds like something along the line of what I’m looking for? Any example of how to configure it? > On 20 Jun 2017, at 19:24, Tihomir Culjaga > wrote: > > mod_rad_auth if you want just to authenticate. > > its an application that you can trigger in dialplan regradless if its an incoming or outgoing call .. > > On 20 June 2017 at 20:22, Joseph Waite > wrote: > There are no users configured on FreeSwitch, all user config is done on JeraSoft VCS and FreeSwitch authenticates via radius! > > FreeSwitch should allow an INVITE from any IP, not send 407, but authenticate via Radius > >> On 20 Jun 2017, at 19:12, Sergey Safarov > wrote: >> >> you can add cidr attribute for all users >> >> вт, 20 июн. 2017 г. в 20:25, Jospeh Waite >: >> I don’t want to allow certain users, I want all calls that come in on this Sofia profile to not send the 407 but still authenticate via Radius based on the IP. >> >> >>> On 20 Jun 2017, at 18:15, Sergey Safarov > wrote: >>> >>> When you generate manually ACL with trusted IP or via mod_xml_radius then you can accept call without 407 message. >>> Also if you add cird attribute to user directory then FreeSwitch can map call to user via cidr attribute. >>> >>> вт, 20 июн. 2017 г. в 19:43, Joseph Waite >: >>> I am using mod_xml_radius, however my issue is that if I enable auth_calls in profile it sends a 407 Proxy Authentication Required sip message, and if I set auth_calls to false it doesn’t authenticate with Radius, it simply passes call straight into the dial plan. >>> >>> >>>> On 20 Jun 2017, at 17:32, Sergey Safarov > wrote: >>>> >>>> You can try mod_xml_radius and generate directory record with cidr adribute. >>>> >>>> >>>> вт, 20 июня 2017 г., 19:25 Joseph Waite >: >>>> Hi Guys >>>> >>>> I am trying to configure a Sofia Profile that will not send a 407 Proxy Authentication Required, but will still authenticate the incoming invite via Radius based on the IP address of the INVITE. >>>> >>>> If I change the Auth_calls to false on the Sofia profile, it doesn’t send the 407, but then it doesn’t authenticate the call. >>>> >>>> Any help would be most appreciated. >>>> >>>> Regards >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From nabeel.10.ahmed at gmail.com Tue Jun 20 18:39:56 2017 From: nabeel.10.ahmed at gmail.com (Nabeel Ahmad) Date: Tue, 20 Jun 2017 23:39:56 +0500 Subject: [Freeswitch-users] Call limit on RTP-IP Address In-Reply-To: <9e2901d2e9dd$391c0eb0$ab542c10$@freeswitch.org> References: <9cad01d2e9ce$3ab34e50$b019eaf0$@freeswitch.org> <5a52f788-d3b1-a3fb-4e0d-daeb247eb782@nevian.org> <9d7b01d2e9d3$366f97e0$a34ec7a0$@freeswitch.org> <9e2901d2e9dd$391c0eb0$ab542c10$@freeswitch.org> Message-ID: Umm...I am on limit page since Morning . limit on any user , outbound gateway all is working well. they are all in dialplan . How can i limit inbound profile this i still can't figure from the link given. And one question for rtp-ip value as i was doing it some other way . If i do like you have mentioned ,sofia replies with this sdp . ** o=FreeSWITCH 1497955081 1497955082 IN IP4 192.168.1.1,192.168.1.2,192.168.1.4 s=FreeSWITCH *c=IN IP4 192.168.1.1,192.168.1.2,192.168.1.4 * I am not sure if this is valid sdp attribute or not. I am doing like this Which on each call offers one ip in sdp like this . o=FreeSWITCH 1497961414 1497961415 IN IP4 192.168.1.3 s=FreeSWITCH c=IN IP4 192.168.1.3 and this on another call . o=FreeSWITCH 1497961414 1497961415 IN IP4 192.168.1.1 s=FreeSWITCH c=IN IP4 192.168.1.1 On Tue, Jun 20, 2017 at 8:52 PM, Ken Rice wrote: > No Defining multiple RTP IPs has been there for a while.... you don’t > define them in different lines, the parser will filter out previous ones, > you define them all together > > > > The stack will then round robin them. This feature was added several > years ago so that FreeSWITCH can handle the required RTP load in traffic > flows that can exceed that of 1Gig-E network connections while only using 1 > SIP Profile for traffic. It's still useful although less of a requirement > with 10GigE coming down in price. (you can now find managed 48port 10GE > switches, NICs and Cables on the secondary market for a combined cost under > $200/port now. > > > -----Original Message----- > From: FreeSWITCH-users [mailto:freeswitch-users- > bounces at lists.freeswitch.org] On Behalf Of Serge S. Yuriev > Sent: Tuesday, June 20, 2017 10:35 AM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Call limit on RTP-IP Address > > I mean if we can define multiple IPs at all. I was under impression parser > uses only one last defined. > > On 20/06/17 17:40, Ken Rice wrote: > > This is not a bug... there is just no way to select which IP or port is > used nor is there an effective way to limit it short of just killing the > call due to the way the IP/Port allocation works in the RTP stack currently. > > > > -----Original Message----- > > From: FreeSWITCH-users > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > > Serge S. Yuriev > > Sent: Tuesday, June 20, 2017 9:17 AM > > To: freeswitch-users at lists.freeswitch.org > > Subject: Re: [Freeswitch-users] Call limit on RTP-IP Address > > > > Is this multi-IP config expected to work this way or it's > eventuality/bug? > > > > On 20/06/17 17:05, Ken Rice wrote: > >> Theres no way to limit or select which IP/Port combination is used > >> from the available RTP IP/Port Range in the config you have. The > >> only way to do this would be to create a profile for each IP and then > >> limit the number of calls per profile > >> > >> *From:* FreeSWITCH-users > >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of > >> *Nabeel Ahmad > >> *Sent:* Tuesday, June 20, 2017 4:16 AM > >> *To:* freeswitch-users at lists.freeswitch.org > >> *Subject:* [Freeswitch-users] Call limit on RTP-IP Address > >> > >> Hello all, > >> > >> If we assign more then one RTP-IP parameter to a sip profile , they > >> are used in round robin. > >> > >> Its perfect . I want to know is there a way to use some limit on that > ip ? > >> > >> Say i've 5 ip address listening on box , and i want one concurrent > >> call limit on each media ip. > >> > >> How can i set limit on profile level or set from dialplan. > >> > >> I tried to do but it didn't work . > >> > >> Any help or advice will be highly appreciated > >> > >> Thanking all > >> > >> Nabeel. > > > > > > -- > > Serge S. Yuriev > > Lead VoIP engineer > > > > > > ______________________________________________________________________ > > ___ Professional FreeSWITCH Consulting Services: > > consulting at 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 at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > > rs > > http://www.freeswitch.org > > > > > > ______________________________________________________________________ > > ___ Professional FreeSWITCH Consulting Services: > > consulting at 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 at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use > > rs > > http://www.freeswitch.org > > > > -- > Serge S. Yuriev > Lead VoIP engineer > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From s.safarov at gmail.com Tue Jun 20 18:44:57 2017 From: s.safarov at gmail.com (Sergey Safarov) Date: Tue, 20 Jun 2017 18:44:57 +0000 Subject: [Freeswitch-users] Radius auth based on IP without sending 407 Proxy authentication required In-Reply-To: References: <84019A4D-1659-496F-8542-BAE10F901641@tm.net.uk> <7FCEE5B4-6B43-4623-ACC1-1709FFBC32A2@tm.net.uk> Message-ID: Mike user IP may be added to ACL and then FreeSwitch will send directory lookup request. вт, 20 июн. 2017 г. в 21:31, Michael Jerris : > the user directory lookup for auth is not done until after we send the > 407. To do this without an acl you need to not do auth on the sip profile, > and handle it in dial plan instead. > > On Jun 20, 2017, at 2:22 PM, Joseph Waite wrote: > > There are no users configured on FreeSwitch, all user config is done on > JeraSoft VCS and FreeSwitch authenticates via radius! > > FreeSwitch should allow an INVITE from any IP, not send 407, but > authenticate via Radius > > On 20 Jun 2017, at 19:12, Sergey Safarov wrote: > > you can add cidr attribute for all users > > вт, 20 июн. 2017 г. в 20:25, Jospeh Waite : > >> I don’t want to allow certain users, I want all calls that come in on >> this Sofia profile to not send the 407 but still authenticate via Radius >> based on the IP. >> >> >> On 20 Jun 2017, at 18:15, Sergey Safarov wrote: >> >> When you generate manually ACL with trusted IP or via mod_xml_radius then >> you can accept call without 407 message. >> Also if you add cird attribute to user directory then FreeSwitch can map >> call to user via cidr attribute. >> >> вт, 20 июн. 2017 г. в 19:43, Joseph Waite : >> >>> I am using mod_xml_radius, however my issue is that if I enable >>> auth_calls in profile it sends a 407 Proxy Authentication Required sip >>> message, and if I set auth_calls to false it doesn’t authenticate with >>> Radius, it simply passes call straight into the dial plan. >>> >>> >>> On 20 Jun 2017, at 17:32, Sergey Safarov wrote: >>> >>> You can try mod_xml_radius and generate directory record with cidr >>> adribute. >>> >>> вт, 20 июня 2017 г., 19:25 Joseph Waite : >>> >>>> Hi Guys >>>> >>>> I am trying to configure a Sofia Profile that will not send a 407 Proxy >>>> Authentication Required, but will still authenticate the incoming invite >>>> via Radius based on the IP address of the INVITE. >>>> >>>> If I change the Auth_calls to false on the Sofia profile, it doesn’t >>>> send the 407, but then it doesn’t authenticate the call. >>>> >>>> Any help would be most appreciated. >>>> >>> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From joelists at tm.net.uk Tue Jun 20 18:46:44 2017 From: joelists at tm.net.uk (Joseph Waite) Date: Tue, 20 Jun 2017 19:46:44 +0100 Subject: [Freeswitch-users] Radius auth based on IP without sending 407 Proxy authentication required In-Reply-To: References: <84019A4D-1659-496F-8542-BAE10F901641@tm.net.uk> <7FCEE5B4-6B43-4623-ACC1-1709FFBC32A2@tm.net.uk> Message-ID: <77718CA2-E233-411A-9983-170DF76B104F@tm.net.uk> One question. the docs in confluence seem to indicate that it can only be used for user/pass authentication and not IP authentication. Is this correct? > On 20 Jun 2017, at 19:28, Tihomir Culjaga wrote: > > i build that module few years ago and it still works as a charm :P > > have fun, > Tihomir. > > On 20 June 2017 at 20:24, Tihomir Culjaga > wrote: > mod_rad_auth if you want just to authenticate. > > its an application that you can trigger in dialplan regradless if its an incoming or outgoing call .. > > On 20 June 2017 at 20:22, Joseph Waite > wrote: > There are no users configured on FreeSwitch, all user config is done on JeraSoft VCS and FreeSwitch authenticates via radius! > > FreeSwitch should allow an INVITE from any IP, not send 407, but authenticate via Radius > >> On 20 Jun 2017, at 19:12, Sergey Safarov > wrote: >> >> you can add cidr attribute for all users >> >> вт, 20 июн. 2017 г. в 20:25, Jospeh Waite >: >> I don’t want to allow certain users, I want all calls that come in on this Sofia profile to not send the 407 but still authenticate via Radius based on the IP. >> >> >>> On 20 Jun 2017, at 18:15, Sergey Safarov > wrote: >>> >>> When you generate manually ACL with trusted IP or via mod_xml_radius then you can accept call without 407 message. >>> Also if you add cird attribute to user directory then FreeSwitch can map call to user via cidr attribute. >>> >>> вт, 20 июн. 2017 г. в 19:43, Joseph Waite >: >>> I am using mod_xml_radius, however my issue is that if I enable auth_calls in profile it sends a 407 Proxy Authentication Required sip message, and if I set auth_calls to false it doesn’t authenticate with Radius, it simply passes call straight into the dial plan. >>> >>> >>>> On 20 Jun 2017, at 17:32, Sergey Safarov > wrote: >>>> >>>> You can try mod_xml_radius and generate directory record with cidr adribute. >>>> >>>> >>>> вт, 20 июня 2017 г., 19:25 Joseph Waite >: >>>> Hi Guys >>>> >>>> I am trying to configure a Sofia Profile that will not send a 407 Proxy Authentication Required, but will still authenticate the incoming invite via Radius based on the IP address of the INVITE. >>>> >>>> If I change the Auth_calls to false on the Sofia profile, it doesn’t send the 407, but then it doesn’t authenticate the call. >>>> >>>> Any help would be most appreciated. >>>> >>>> Regards >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From ahmed at netelsat.net Tue Jun 20 18:54:25 2017 From: ahmed at netelsat.net (Ahmed Sboor) Date: Tue, 20 Jun 2017 23:54:25 +0500 Subject: [Freeswitch-users] Radius auth based on IP without sending 407 Proxy authentication required In-Reply-To: <77718CA2-E233-411A-9983-170DF76B104F@tm.net.uk> References: <84019A4D-1659-496F-8542-BAE10F901641@tm.net.uk> <7FCEE5B4-6B43-4623-ACC1-1709FFBC32A2@tm.net.uk> <77718CA2-E233-411A-9983-170DF76B104F@tm.net.uk> Message-ID: Hi Joseph, if you are using mod_xml_radius , i am not sure how your Dialplan is , but if you are not missing you have to include file from mod_xml_radius source in your dialplan. 00_dialplan_auth.xml When we were using VCS , both ip/user authentication worked well . On Tue, Jun 20, 2017 at 11:46 PM, Joseph Waite wrote: > One question. the docs in confluence seem to indicate that it can only be > used for user/pass authentication and not IP authentication. Is this > correct? > > On 20 Jun 2017, at 19:28, Tihomir Culjaga wrote: > > i build that module few years ago and it still works as a charm :P > > have fun, > Tihomir. > > On 20 June 2017 at 20:24, Tihomir Culjaga wrote: > >> mod_rad_auth if you want just to authenticate. >> >> its an application that you can trigger in dialplan regradless if its an >> incoming or outgoing call .. >> >> On 20 June 2017 at 20:22, Joseph Waite wrote: >> >>> There are no users configured on FreeSwitch, all user config is done on >>> JeraSoft VCS and FreeSwitch authenticates via radius! >>> >>> FreeSwitch should allow an INVITE from any IP, not send 407, but >>> authenticate via Radius >>> >>> On 20 Jun 2017, at 19:12, Sergey Safarov wrote: >>> >>> you can add cidr attribute for all users >>> >>> вт, 20 июн. 2017 г. в 20:25, Jospeh Waite : >>> >>>> I don’t want to allow certain users, I want all calls that come in on >>>> this Sofia profile to not send the 407 but still authenticate via Radius >>>> based on the IP. >>>> >>>> >>>> On 20 Jun 2017, at 18:15, Sergey Safarov wrote: >>>> >>>> When you generate manually ACL with trusted IP or via mod_xml_radius >>>> then you can accept call without 407 message. >>>> Also if you add cird attribute to user directory then FreeSwitch can >>>> map call to user via cidr attribute. >>>> >>>> вт, 20 июн. 2017 г. в 19:43, Joseph Waite : >>>> >>>>> I am using mod_xml_radius, however my issue is that if I enable >>>>> auth_calls in profile it sends a 407 Proxy Authentication Required sip >>>>> message, and if I set auth_calls to false it doesn’t authenticate with >>>>> Radius, it simply passes call straight into the dial plan. >>>>> >>>>> >>>>> On 20 Jun 2017, at 17:32, Sergey Safarov wrote: >>>>> >>>>> You can try mod_xml_radius and generate directory record with cidr >>>>> adribute. >>>>> >>>>> вт, 20 июня 2017 г., 19:25 Joseph Waite : >>>>> >>>>>> Hi Guys >>>>>> >>>>>> I am trying to configure a Sofia Profile that will not send a 407 >>>>>> Proxy Authentication Required, but will still authenticate the incoming >>>>>> invite via Radius based on the IP address of the INVITE. >>>>>> >>>>>> If I change the Auth_calls to false on the Sofia profile, it doesn’t >>>>>> send the 407, but then it doesn’t authenticate the call. >>>>>> >>>>>> Any help would be most appreciated. >>>>>> >>>>>> Regards >>>>>> >>>>>> ____________________________________________________________ >>>>>> _____________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at 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 at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>>>> switch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>> ____________________________________________________________ >>>>> _____________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at 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 at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>>> switch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> ____________________________________________________________ >>>>> _____________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at 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 at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>>> switch-users >>>>> http://www.freeswitch.org >>>>> >>>> ____________________________________________________________ >>>> _____________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at 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 at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>> switch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> ____________________________________________________________ >>>> _____________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at 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 at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>> switch-users >>>> http://www.freeswitch.org >>>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at 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 at 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 at 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 at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From joelists at tm.net.uk Tue Jun 20 19:01:10 2017 From: joelists at tm.net.uk (Jospeh Waite) Date: Tue, 20 Jun 2017 20:01:10 +0100 Subject: [Freeswitch-users] Routing Calls to registered endpoints. Message-ID: Hi Guys. Final question of the day. I promise!! I have 2 sofia profiles, one for IP authenticated calls and one for Sip Registrations. There both on port 5060 on 2 different IP addresses. This is basically a Wholesale/retail billing/switching platform. We have a JeraSoft VCS radius based billing and Sip Redirect server. Everything works ok, except for inbound calls from DID provider destined for a registered extension. Call comes in on the IP profile, which then sends a request to the JeraSoft SIP Redirect server, which replies with a 300 multiple choices as follows. SIP/2.0 300 Multiple Choices Via: SIP/2.0/UDP {ip of FS IP profile};rport;branch=z9hG4bKSKFBFy6K81U9H From: "07966677711" ;tag=S6Xmmj7meS2Fr To: Contact: ;q=1.00 Call-ID: aed22c59-d082-1235-baa2-363165383663 CSeq: 108654212 INVITE Max-Forwards: 67 Content-Length: 0 Server: JeraSoft VCS SIP Redirect Server joehouse is the username of a registered user, registered to the reg profile. Now FS instead of connecting the call to the registered user it sends a new request to the Jerasoft redirect. How would I get it to route the call to the registered user? public/30_routing.xml --> public.xml -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Tue Jun 20 19:03:15 2017 From: mike at jerris.com (Michael Jerris) Date: Tue, 20 Jun 2017 15:03:15 -0400 Subject: [Freeswitch-users] Radius auth based on IP without sending 407 Proxy authentication required In-Reply-To: References: <84019A4D-1659-496F-8542-BAE10F901641@tm.net.uk> <7FCEE5B4-6B43-4623-ACC1-1709FFBC32A2@tm.net.uk> Message-ID: Yes, but he’s specifically said he doesn’t have the users in dir… this option has already been discussed, so what I pointed out is the only other option, and the only way to do this totally dynamically. > On Jun 20, 2017, at 2:44 PM, Sergey Safarov wrote: > > Mike user IP may be added to ACL and then FreeSwitch will send directory lookup request. > > вт, 20 июн. 2017 г. в 21:31, Michael Jerris >: > the user directory lookup for auth is not done until after we send the 407. To do this without an acl you need to not do auth on the sip profile, and handle it in dial plan instead. > >> On Jun 20, 2017, at 2:22 PM, Joseph Waite > wrote: >> >> There are no users configured on FreeSwitch, all user config is done on JeraSoft VCS and FreeSwitch authenticates via radius! >> >> FreeSwitch should allow an INVITE from any IP, not send 407, but authenticate via Radius >>> On 20 Jun 2017, at 19:12, Sergey Safarov > wrote: >>> >>> you can add cidr attribute for all users >>> >>> вт, 20 июн. 2017 г. в 20:25, Jospeh Waite >: >>> I don’t want to allow certain users, I want all calls that come in on this Sofia profile to not send the 407 but still authenticate via Radius based on the IP. >>> >>> >>>> On 20 Jun 2017, at 18:15, Sergey Safarov > wrote: >>>> >>>> When you generate manually ACL with trusted IP or via mod_xml_radius then you can accept call without 407 message. >>>> Also if you add cird attribute to user directory then FreeSwitch can map call to user via cidr attribute. >>>> >>>> вт, 20 июн. 2017 г. в 19:43, Joseph Waite >: >>>> I am using mod_xml_radius, however my issue is that if I enable auth_calls in profile it sends a 407 Proxy Authentication Required sip message, and if I set auth_calls to false it doesn’t authenticate with Radius, it simply passes call straight into the dial plan. >>>> >>>> >>>>> On 20 Jun 2017, at 17:32, Sergey Safarov > wrote: >>>>> >>>>> You can try mod_xml_radius and generate directory record with cidr adribute. >>>>> >>>>> >>>>> вт, 20 июня 2017 г., 19:25 Joseph Waite >: >>>>> Hi Guys >>>>> >>>>> I am trying to configure a Sofia Profile that will not send a 407 Proxy Authentication Required, but will still authenticate the incoming invite via Radius based on the IP address of the INVITE. >>>>> >>>>> If I change the Auth_calls to false on the Sofia profile, it doesn’t send the 407, but then it doesn’t authenticate the call. >>>>> >>>>> Any help would be most appreciated. > -------------- next part -------------- An HTML attachment was scrubbed... URL: From anthony.minessale at gmail.com Tue Jun 20 22:01:50 2017 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 20 Jun 2017 22:01:50 +0000 Subject: [Freeswitch-users] Call limit on RTP-IP Address In-Reply-To: References: <9cad01d2e9ce$3ab34e50$b019eaf0$@freeswitch.org> <5a52f788-d3b1-a3fb-4e0d-daeb247eb782@nevian.org> <9d7b01d2e9d3$366f97e0$a34ec7a0$@freeswitch.org> <9e2901d2e9dd$391c0eb0$ab542c10$@freeswitch.org> Message-ID: Its not possible. Maybe file a bounty. On Tue, Jun 20, 2017 at 2:44 PM Nabeel Ahmad wrote: > Umm...I am on limit page since Morning . limit on any user , outbound > gateway all is working well. they are all in dialplan . How can i limit > inbound profile this i still can't figure from the link given. > > And one question for rtp-ip value as i was doing it some other way . > If i do like you have mentioned ,sofia replies with this sdp . > > ** > > o=FreeSWITCH 1497955081 1497955082 IN IP4 > 192.168.1.1,192.168.1.2,192.168.1.4 > s=FreeSWITCH > *c=IN IP4 192.168.1.1,192.168.1.2,192.168.1.4 * > > I am not sure if this is valid sdp attribute or not. > > I am doing like this > > > > > Which on each call offers one ip in sdp like this . > o=FreeSWITCH 1497961414 1497961415 IN IP4 192.168.1.3 > s=FreeSWITCH > c=IN IP4 192.168.1.3 > > and this on another call . > > o=FreeSWITCH 1497961414 1497961415 IN IP4 192.168.1.1 > s=FreeSWITCH > c=IN IP4 192.168.1.1 > > > > On Tue, Jun 20, 2017 at 8:52 PM, Ken Rice wrote: > >> No Defining multiple RTP IPs has been there for a while.... you don’t >> define them in different lines, the parser will filter out previous ones, >> you define them all together >> >> >> >> The stack will then round robin them. This feature was added several >> years ago so that FreeSWITCH can handle the required RTP load in traffic >> flows that can exceed that of 1Gig-E network connections while only using 1 >> SIP Profile for traffic. It's still useful although less of a requirement >> with 10GigE coming down in price. (you can now find managed 48port 10GE >> switches, NICs and Cables on the secondary market for a combined cost under >> $200/port now. >> >> >> -----Original Message----- >> From: FreeSWITCH-users [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Serge S. >> Yuriev >> Sent: Tuesday, June 20, 2017 10:35 AM >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] Call limit on RTP-IP Address >> >> I mean if we can define multiple IPs at all. I was under impression >> parser uses only one last defined. >> >> On 20/06/17 17:40, Ken Rice wrote: >> > This is not a bug... there is just no way to select which IP or port is >> used nor is there an effective way to limit it short of just killing the >> call due to the way the IP/Port allocation works in the RTP stack currently. >> > >> > -----Original Message----- >> > From: FreeSWITCH-users >> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of >> > Serge S. Yuriev >> > Sent: Tuesday, June 20, 2017 9:17 AM >> > To: freeswitch-users at lists.freeswitch.org >> > Subject: Re: [Freeswitch-users] Call limit on RTP-IP Address >> > >> > Is this multi-IP config expected to work this way or it's >> eventuality/bug? >> > >> > On 20/06/17 17:05, Ken Rice wrote: >> >> Theres no way to limit or select which IP/Port combination is used >> >> from the available RTP IP/Port Range in the config you have. The >> >> only way to do this would be to create a profile for each IP and then >> >> limit the number of calls per profile >> >> >> >> *From:* FreeSWITCH-users >> >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of >> >> *Nabeel Ahmad >> >> *Sent:* Tuesday, June 20, 2017 4:16 AM >> >> *To:* freeswitch-users at lists.freeswitch.org >> >> *Subject:* [Freeswitch-users] Call limit on RTP-IP Address >> >> >> >> Hello all, >> >> >> >> If we assign more then one RTP-IP parameter to a sip profile , they >> >> are used in round robin. >> >> >> >> Its perfect . I want to know is there a way to use some limit on that >> ip ? >> >> >> >> Say i've 5 ip address listening on box , and i want one concurrent >> >> call limit on each media ip. >> >> >> >> How can i set limit on profile level or set from dialplan. >> >> >> >> I tried to do but it didn't work . >> >> >> >> Any help or advice will be highly appreciated >> >> >> >> Thanking all >> >> >> >> Nabeel. >> > >> > >> > -- >> > Serge S. Yuriev >> > Lead VoIP engineer >> > >> > >> > ______________________________________________________________________ >> > ___ Professional FreeSWITCH Consulting Services: >> > consulting at 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 at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use >> > rs >> > http://www.freeswitch.org >> > >> > >> > ______________________________________________________________________ >> > ___ Professional FreeSWITCH Consulting Services: >> > consulting at 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 at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use >> > rs >> > http://www.freeswitch.org >> > >> >> -- >> Serge S. Yuriev >> Lead VoIP engineer >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at 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 at 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 at 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 at 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 at 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 ♬ @anthmfs ♬ @FreeSWITCH ♬ ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ * ClueCon Weekly Development Call ☎ sip:888 at conference.freeswitch.org ☎ +19193869900 https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA -------------- next part -------------- An HTML attachment was scrubbed... URL: From schoch+freeswitch.org at xwin32.com Tue Jun 20 22:55:28 2017 From: schoch+freeswitch.org at xwin32.com (Steven Schoch) Date: Tue, 20 Jun 2017 15:55:28 -0700 Subject: [Freeswitch-users] IVR menu-back after a timeout Message-ID: Is it possible for an IVR (in XML) to do a "menu-back" to go back to the top-level menu after a timeout? -- Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: From yehavi.bourvine at gmail.com Wed Jun 21 06:46:16 2017 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Wed, 21 Jun 2017 09:46:16 +0300 Subject: [Freeswitch-users] lua os.execute fails after Centos 7.3 update (21-Jun-2017) Message-ID: Hello, Due to the recent security issues I did an update to my centos 7.3 system which runs Freeswitch 1.6.17/18 (two systems); The main packages updated were the kernel and Libc. Since then, os.execute stopped working inside lua called scripts (luarun api); they fail with error 32512 which means that the command is not available. However, before the upgrade it worked, and so works the command itself when I run it from bash. Any idea? Some permissions has been changed at the security update? Thanks, __Yehavi: -------------- next part -------------- An HTML attachment was scrubbed... URL: From agubbe at gmail.com Wed Jun 21 07:30:44 2017 From: agubbe at gmail.com (=?UTF-8?Q?Agust=C3=AD_Ubalde_Bellot?=) Date: Wed, 21 Jun 2017 09:30:44 +0200 Subject: [Freeswitch-users] Freeswitch sslv3 support In-Reply-To: References: Message-ID: Hi Michael, I've created this ISSUE: https://freeswitch.org/jira/browse/FS-10406. Regards, Agustí 2017-06-20 17:55 GMT+02:00 Agustí Ubalde Bellot : > Hi Michael, > > Yes, the version I am using is a development version (1.5.14). In any > case, I have performed the same tests in version 1.6 and have the same > behavior. > Instead, the verto module does block the sslv3 protocol. > > > Thanks, > Agustí > > 2017-06-20 10:45 GMT+02:00 Agustí Ubalde Bellot : > >> Hi Michael, >> >> I have performed several connection tests forcing the sslv3 protocol over >> secure web sockets and the connection is established. Instead, the same >> test connecting to the TLS listening port, the connection is not set. The >> protocol is successfully disabled in the configuration. >> The version of FreeSWITCH I'm testing is 1.5.14. Is there any way to >> prove that the sslv3 protocol is actually disabled in this release? >> >> >> Thanks, >> Agustí >> >> 2017-06-15 10:07 GMT+02:00 Agustí Ubalde Bellot : >> >>> Hi Brian, >>> >>> Is possible to disable for web socket secure connections too? >>> >>> >>> Thanks, >>> Agustí >>> >>> 2017-06-13 13:24 GMT+02:00 Agustí Ubalde Bellot : >>> >>>> Hi all, >>>> >>>> Is there a FreeSWITCH update where sslv3 support is disabled? >>>> >>>> >>>> Thanks, >>>> Agustí >>>> >>> >>> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: From gmaruzz at gmail.com Wed Jun 21 07:45:00 2017 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Wed, 21 Jun 2017 09:45:00 +0200 Subject: [Freeswitch-users] IVR menu-back after a timeout In-Reply-To: References: Message-ID: On 21 June 2017 at 00:55, Steven Schoch wrote: > Is it possible for an IVR (in XML) to do a "menu-back" to go back to the > top-level menu after a timeout? > > nope -------------- next part -------------- An HTML attachment was scrubbed... URL: From achinthau at gmail.com Wed Jun 21 10:54:37 2017 From: achinthau at gmail.com (Achintha) Date: Wed, 21 Jun 2017 16:24:37 +0530 Subject: [Freeswitch-users] switch_core_sqldb.c:2987 invalied cdr data, call not recoverd In-Reply-To: <5A5DBC29-DCE6-417F-8B48-2BE9A3390523@jerris.com> References: <5A5DBC29-DCE6-417F-8B48-2BE9A3390523@jerris.com> Message-ID: hi michael thanks a lot for your quick respond here i attached the logs of both freeswitch servers and params files. testing scenario is first i put a call to fs_master and then crashed it. Then the call landed to fs_slave but RTP did not function. i attatched following files and also put it on pastebin https://pastebin.freeswitch.org/view/c29ac12a On Tue, Jun 20, 2017 at 9:14 PM, Michael Jerris wrote: > would need a bug report on this one with full logs and config and how to > reproduce to look into it. > > On Jun 20, 2017, at 11:05 AM, Achintha wrote: > > hi all, > > I configured two freeswitch servers (freeswitch 1.6.18 on debian 8 ) with > call recovery feature. It is working properly on extension to extension , > out bound and IVR Calls. > Then i tried to land an incoming call, the call gets connected to the > queue and then freeswitch generate a call to agent and bridge it with > queued call and both sides can hear properly. > Then i crashed the primary server, call got landed to the second > freeswitch server but rtp is not functioning, and in the second freeswitch > console, it printed "switch_core_sqldb.c:2987 invalied cdr data, call not > recovered". But the call was not disconnected. > > I used the following configurations > > *switch.conf.xml : * > > switch name is same on both servers > core-db-dsn and core-recovery-db-dsn configured > with pgsql > *both sip profiles:* > odbc-dsn configured with pgsql > track-calls elabled > > please provide me a solution to sort out this. > > -- > Best Regards.. > Achintha Udukumbura > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Best Regards.. Achintha Udukumbura -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: fs_master.log Type: application/octet-stream Size: 42814 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: fs_slave.log Type: application/octet-stream Size: 11923 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: external.xml Type: text/xml Size: 5188 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: internal.xml Type: text/xml Size: 21311 bytes Desc: not available URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: switch.conf.xml Type: text/xml Size: 8010 bytes Desc: not available URL: From italo at freeswitch.org Wed Jun 21 12:44:29 2017 From: italo at freeswitch.org (=?UTF-8?B?w410YWxvIFJvc3Np?=) Date: Wed, 21 Jun 2017 09:44:29 -0300 Subject: [Freeswitch-users] Callcenter module, can I originate call for an agent in uuid-standby mode ? In-Reply-To: <8C4372DB-F2EB-4614-82EC-185B5AA33C65@jerris.com> References: <4770F7F5-BCBB-47B6-A909-263EEFEADFE4@jerris.com> <1497890891.572062.1014326656.673B64EF@webmail.messagingengine.com> <8C4372DB-F2EB-4614-82EC-185B5AA33C65@jerris.com> Message-ID: uuid-standby is the same as a Consumer agent for mod_fifo (off-hook agent). Khalil, Yes it's possible, it's just a matter of putting the right commands on your dialplan. When you log in your uuid-standby agent, put the session uuid in a hash (hash insert/agents_uuid/myagent/uuid), then you query this hash (hash select/agents_uuid/myagent) before originate and set a variable on the originate channel, when the call is established you call intercept app with the agent uuid: This will bridge your originated call with the agent's standby leg. You can even pause the agent before calling intercept to make sure the agent doesn't get another call or call callcenter_track app callcenter_track(myagent), this will tell mod_callcenter that this agent has an external call and the agent won't be called until the number of external_calls is 0 again. I'm pretty sure you can do the same with mod_fifo. On Tue, Jun 20, 2017 at 12:43 PM, Michael Jerris wrote: > Not sure what that means exactly. > > On Jun 20, 2017, at 10:00 AM, Khalil Khamlichi > wrote: > > alright, does mod_fifo support some sort of uuid_standby mode ? > > On Mon, Jun 19, 2017 at 6:27 PM, Michael Jerris wrote: > >> mod_fifo has ALWAYS been superior, people assume otherwise because of the >> name. Check it out, its pretty powerful. mod_callcenter was written >> because people had a hard time understanding mod_fifo. It supports agent >> tracking, some skills routing, inbound and outbound agents, etc. If there >> is stuff missing we should really sort getting it into mod_fifo and abandon >> mod_callcenter. >> >> >> > On Jun 19, 2017, at 12:48 PM, Michael Avers >> wrote: >> > >> > What makes mod_fifo better these days? Some reason I was under the >> impression mod_callcenter is a better choice. Can you please give some real >> world examples where mod_fifo excels compared to mod_callcenter, or >> features that are not possible to implement with the latter? >> > >> > Thanks >> > Mike >> > >> > On Mon, Jun 19, 2017, at 09:34 AM, Michael Jerris wrote: >> >> mod_fifo is a much more feature rich version of a call queue than >> mod_callcenter. You might want to check that out instead. >> >> >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Ítalo Rossi italo at freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From italo at freeswitch.org Wed Jun 21 12:45:51 2017 From: italo at freeswitch.org (=?UTF-8?B?w410YWxvIFJvc3Np?=) Date: Wed, 21 Jun 2017 09:45:51 -0300 Subject: [Freeswitch-users] Calling on the community for Bug Marshals In-Reply-To: References: Message-ID: Cool! Can you guys join our hipchat ? There's a channel there called Bughunt, join there and we can chat about. On Tue, Jun 20, 2017 at 1:35 PM, wrote: > I would be happy to help. Let me know what the "requirements" are. > > Sent from my iPhone > > On Jun 20, 2017, at 10:30 AM, David Villasmil com> wrote: > > Anything i can help with? > On Tue, Jun 20, 2017 at 2:46 PM Ítalo Rossi wrote: > >> This is a great opportunity to learn and to be an expert in FreeSWITCH. >> This was how I learn a lot! >> >> :-) >> >> On Mon, Jun 5, 2017 at 4:39 PM, Brian West wrote: >> >>> FreeSWITCHers, >>> >>> We are in need of a few good bug marshals, We are trying to get 1.8 >>> ready and out the door and the more help we have testing and working thru >>> patches on JIRA the quicker it will arrive. If you're interested in >>> helping us out email me directly. We are also considering bringing back a >>> few days a week we are sitting in 888 and helping the community out with >>> issues pending in JIRA. >>> >>> Also we are only about 2600 short on the gofund me for the Allison >>> prompts, which will be delivered sometime this week. ;) So help us get >>> over that last little bit this week. >>> >>> Thanks, >>> >>> -- >>> >>> *Brian West* >>> brian at freeswitch.org >>> >>> *Twitter: @FreeSWITCH , @briankwest* >>> >>> http://www.freeswitchbook.com >>> http://www.freeswitchcookbook.com >>> >>> Book a phone call (CST) >>> >>> Allison prompts for FreeSWITCH: >>> >>> *https://www.gofundme.com/allison-prompts-for-freeswitch* >>> >>> >>> Got Bugs? Report them here ! | Reddit: >>> /r/freeswitch >>> >>> *T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002 >>> <(918)%20420-9002> | *M:*+1918424WEST (9378) >>> *Skype:*briankwest >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at 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 at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Ítalo Rossi >> italo at freeswitch.org >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Ítalo Rossi italo at freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From luis.daniel.lucio at gmail.com Wed Jun 21 11:59:10 2017 From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz) Date: Wed, 21 Jun 2017 07:59:10 -0400 Subject: [Freeswitch-users] lua os.execute fails after Centos 7.3 update (21-Jun-2017) In-Reply-To: References: Message-ID: Try turning off selinux Le 21 juin 2017 2:47 AM, "Yehavi Bourvine" a écrit : > Hello, > > Due to the recent security issues I did an update to my centos 7.3 > system which runs Freeswitch 1.6.17/18 (two systems); The main packages > updated were the kernel and Libc. Since then, os.execute stopped working > inside lua called scripts (luarun api); they fail with error 32512 which > means that the command is not available. However, before the upgrade it > worked, and so works the command itself when I run it from bash. > > Any idea? Some permissions has been changed at the security update? > > Thanks, __Yehavi: > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From khamlichi.khalil at gmail.com Wed Jun 21 10:35:18 2017 From: khamlichi.khalil at gmail.com (Khalil Khamlichi) Date: Wed, 21 Jun 2017 10:35:18 +0000 Subject: [Freeswitch-users] Callcenter module, can I originate call for an agent in uuid-standby mode ? In-Reply-To: <8C4372DB-F2EB-4614-82EC-185B5AA33C65@jerris.com> References: <4770F7F5-BCBB-47B6-A909-263EEFEADFE4@jerris.com> <1497890891.572062.1014326656.673B64EF@webmail.messagingengine.com> <8C4372DB-F2EB-4614-82EC-185B5AA33C65@jerris.com> Message-ID: when a customer calls, freeswitch has two ways to reach an agent 1 - dial the agent extension and hand over the customer to him or 2 - if agent is already on a call with freeswitch (using uuid-mode of callcenter module) just bridge customer to this session. so my question is , does mod_fifo has this same functionality in method number 2 of reaching agent ? On Tue, Jun 20, 2017 at 3:43 PM, Michael Jerris wrote: > Not sure what that means exactly. > > On Jun 20, 2017, at 10:00 AM, Khalil Khamlichi > wrote: > > alright, does mod_fifo support some sort of uuid_standby mode ? > > On Mon, Jun 19, 2017 at 6:27 PM, Michael Jerris wrote: > >> mod_fifo has ALWAYS been superior, people assume otherwise because of the >> name. Check it out, its pretty powerful. mod_callcenter was written >> because people had a hard time understanding mod_fifo. It supports agent >> tracking, some skills routing, inbound and outbound agents, etc. If there >> is stuff missing we should really sort getting it into mod_fifo and abandon >> mod_callcenter. >> >> >> > On Jun 19, 2017, at 12:48 PM, Michael Avers >> wrote: >> > >> > What makes mod_fifo better these days? Some reason I was under the >> impression mod_callcenter is a better choice. Can you please give some real >> world examples where mod_fifo excels compared to mod_callcenter, or >> features that are not possible to implement with the latter? >> > >> > Thanks >> > Mike >> > >> > On Mon, Jun 19, 2017, at 09:34 AM, Michael Jerris wrote: >> >> mod_fifo is a much more feature rich version of a call queue than >> mod_callcenter. You might want to check that out instead. >> >> >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From stefan at fuhrmann.homedns.org Wed Jun 21 12:00:44 2017 From: stefan at fuhrmann.homedns.org (Stefan Fuhrmann) Date: Wed, 21 Jun 2017 14:00:44 +0200 Subject: [Freeswitch-users] enable Portal, error 404 In-Reply-To: References: <8779966.9K6fzknjRZ@stefan-ubu> <1794751.l0UOKoQrpU@stefan-ubu> Message-ID: <2151629.A0dYCELd5T@stefan-ubu> Am Dienstag, 20. Juni 2017, 14:31:40 CEST schrieb David Villasmil: > Does netstat shows the port in use? yes, netstat shows it and I can login. netstat -tulpn tcp 0 0 0.0.0.0:8080 0.0.0.0:* LISTEN 718/freeswitch After login the 404 comes up. Tia Stefan From yehavi.bourvine at gmail.com Wed Jun 21 13:37:51 2017 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Wed, 21 Jun 2017 16:37:51 +0300 Subject: [Freeswitch-users] lua os.execute fails after Centos 7.3 update (21-Jun-2017) In-Reply-To: References: Message-ID: It is already disabled. I've found a new thread on RedHat's site that indeed the new patch breaks various type of process creation syscalls (like exec, system(), etc.). I tried raising the stack size as they suggest, but it does not help. Thanks, __Yehavi: 2017-06-21 14:59 GMT+03:00 Luis Daniel Lucio Quiroz < luis.daniel.lucio at gmail.com>: > Try turning off selinux > > Le 21 juin 2017 2:47 AM, "Yehavi Bourvine" a > écrit : > >> Hello, >> >> Due to the recent security issues I did an update to my centos 7.3 >> system which runs Freeswitch 1.6.17/18 (two systems); The main packages >> updated were the kernel and Libc. Since then, os.execute stopped working >> inside lua called scripts (luarun api); they fail with error 32512 which >> means that the command is not available. However, before the upgrade it >> worked, and so works the command itself when I run it from bash. >> >> Any idea? Some permissions has been changed at the security update? >> >> Thanks, __Yehavi: >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From yehavi.bourvine at gmail.com Wed Jun 21 14:22:11 2017 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Wed, 21 Jun 2017 17:22:11 +0300 Subject: [Freeswitch-users] lua os.execute fails after Centos 7.3 update (21-Jun-2017) In-Reply-To: References: Message-ID: Hi, I managed to find a bypass in the meantime; see FS-10410. I have no idea what are the side effects of it, but at least I can work now. BTW, it looks that reports about other software packages being broken by the security patch start to publish... Regards, __Yehavi: 2017-06-21 16:37 GMT+03:00 Yehavi Bourvine : > It is already disabled. I've found a new thread on RedHat's site that > indeed the new patch breaks various type of process creation syscalls (like > exec, system(), etc.). I tried raising the stack size as they suggest, but > it does not help. > > Thanks, __Yehavi: > > 2017-06-21 14:59 GMT+03:00 Luis Daniel Lucio Quiroz < > luis.daniel.lucio at gmail.com>: > >> Try turning off selinux >> >> Le 21 juin 2017 2:47 AM, "Yehavi Bourvine" a >> écrit : >> >>> Hello, >>> >>> Due to the recent security issues I did an update to my centos 7.3 >>> system which runs Freeswitch 1.6.17/18 (two systems); The main packages >>> updated were the kernel and Libc. Since then, os.execute stopped working >>> inside lua called scripts (luarun api); they fail with error 32512 which >>> means that the command is not available. However, before the upgrade it >>> worked, and so works the command itself when I run it from bash. >>> >>> Any idea? Some permissions has been changed at the security update? >>> >>> Thanks, __Yehavi: >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at 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 at 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 at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From joel at gogii.net Wed Jun 21 15:13:57 2017 From: joel at gogii.net (Joel Serrano) Date: Wed, 21 Jun 2017 08:13:57 -0700 Subject: [Freeswitch-users] Calling on the community for Bug Marshals In-Reply-To: References: Message-ID: What is the server name for FS's Hipchat? On Wed, Jun 21, 2017 at 5:45 AM, Ítalo Rossi wrote: > Cool! > > Can you guys join our hipchat ? There's a channel there called Bughunt, > join there and we can chat about. > > On Tue, Jun 20, 2017 at 1:35 PM, wrote: > >> I would be happy to help. Let me know what the "requirements" are. >> >> Sent from my iPhone >> >> On Jun 20, 2017, at 10:30 AM, David Villasmil < >> david.villasmil.work at gmail.com> wrote: >> >> Anything i can help with? >> On Tue, Jun 20, 2017 at 2:46 PM Ítalo Rossi wrote: >> >>> This is a great opportunity to learn and to be an expert in FreeSWITCH. >>> This was how I learn a lot! >>> >>> :-) >>> >>> On Mon, Jun 5, 2017 at 4:39 PM, Brian West wrote: >>> >>>> FreeSWITCHers, >>>> >>>> We are in need of a few good bug marshals, We are trying to get 1.8 >>>> ready and out the door and the more help we have testing and working thru >>>> patches on JIRA the quicker it will arrive. If you're interested in >>>> helping us out email me directly. We are also considering bringing back a >>>> few days a week we are sitting in 888 and helping the community out with >>>> issues pending in JIRA. >>>> >>>> Also we are only about 2600 short on the gofund me for the Allison >>>> prompts, which will be delivered sometime this week. ;) So help us get >>>> over that last little bit this week. >>>> >>>> Thanks, >>>> >>>> -- >>>> >>>> *Brian West* >>>> brian at freeswitch.org >>>> >>>> *Twitter: @FreeSWITCH , @briankwest* >>>> >>>> http://www.freeswitchbook.com >>>> http://www.freeswitchcookbook.com >>>> >>>> Book a phone call (CST) >>>> >>>> Allison prompts for FreeSWITCH: >>>> >>>> *https://www.gofundme.com/allison-prompts-for-freeswitch* >>>> >>>> >>>> Got Bugs? Report them here ! | Reddit: >>>> /r/freeswitch >>>> >>>> *T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002 >>>> <(918)%20420-9002> | *M:*+1918424WEST (9378) >>>> *Skype:*briankwest >>>> >>>> ____________________________________________________________ >>>> _____________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at 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 at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ >>>> freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> >>> -- >>> Ítalo Rossi >>> italo at freeswitch.org >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at 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 at 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 at 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 at 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 at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Ítalo Rossi > italo at freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From italo at freeswitch.org Wed Jun 21 15:32:16 2017 From: italo at freeswitch.org (=?UTF-8?B?w410YWxvIFJvc3Np?=) Date: Wed, 21 Jun 2017 12:32:16 -0300 Subject: [Freeswitch-users] Calling on the community for Bug Marshals In-Reply-To: References: Message-ID: hipchat.freeswitch.org On Wed, Jun 21, 2017 at 12:13 PM, Joel Serrano wrote: > What is the server name for FS's Hipchat? > > On Wed, Jun 21, 2017 at 5:45 AM, Ítalo Rossi wrote: > >> Cool! >> >> Can you guys join our hipchat ? There's a channel there called Bughunt, >> join there and we can chat about. >> >> On Tue, Jun 20, 2017 at 1:35 PM, wrote: >> >>> I would be happy to help. Let me know what the "requirements" are. >>> >>> Sent from my iPhone >>> >>> On Jun 20, 2017, at 10:30 AM, David Villasmil < >>> david.villasmil.work at gmail.com> wrote: >>> >>> Anything i can help with? >>> On Tue, Jun 20, 2017 at 2:46 PM Ítalo Rossi >>> wrote: >>> >>>> This is a great opportunity to learn and to be an expert in FreeSWITCH. >>>> This was how I learn a lot! >>>> >>>> :-) >>>> >>>> On Mon, Jun 5, 2017 at 4:39 PM, Brian West >>>> wrote: >>>> >>>>> FreeSWITCHers, >>>>> >>>>> We are in need of a few good bug marshals, We are trying to get 1.8 >>>>> ready and out the door and the more help we have testing and working thru >>>>> patches on JIRA the quicker it will arrive. If you're interested in >>>>> helping us out email me directly. We are also considering bringing back a >>>>> few days a week we are sitting in 888 and helping the community out with >>>>> issues pending in JIRA. >>>>> >>>>> Also we are only about 2600 short on the gofund me for the Allison >>>>> prompts, which will be delivered sometime this week. ;) So help us get >>>>> over that last little bit this week. >>>>> >>>>> Thanks, >>>>> >>>>> -- >>>>> >>>>> *Brian West* >>>>> brian at freeswitch.org >>>>> >>>>> *Twitter: @FreeSWITCH , @briankwest* >>>>> >>>>> http://www.freeswitchbook.com >>>>> http://www.freeswitchcookbook.com >>>>> >>>>> Book a phone call (CST) >>>>> >>>>> Allison prompts for FreeSWITCH: >>>>> >>>>> *https://www.gofundme.com/allison-prompts-for-freeswitch* >>>>> >>>>> >>>>> Got Bugs? Report them here ! | Reddit: >>>>> /r/freeswitch >>>>> >>>>> *T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002 >>>>> <(918)%20420-9002> | *M:*+1918424WEST (9378) >>>>> *Skype:*briankwest >>>>> >>>>> ____________________________________________________________ >>>>> _____________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at 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 at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>>> switch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> >>>> -- >>>> Ítalo Rossi >>>> italo at freeswitch.org >>>> ____________________________________________________________ >>>> _____________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at 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 at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>> switch-users >>>> http://www.freeswitch.org >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at 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 at 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 at 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 at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Ítalo Rossi >> italo at freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Ítalo Rossi italo at freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Wed Jun 21 15:44:15 2017 From: mike at jerris.com (Michael Jerris) Date: Wed, 21 Jun 2017 11:44:15 -0400 Subject: [Freeswitch-users] switch_core_sqldb.c:2987 invalied cdr data, call not recoverd In-Reply-To: References: <5A5DBC29-DCE6-417F-8B48-2BE9A3390523@jerris.com> Message-ID: Bug reports get filed at https://freeswitch.org/jira not here > On Jun 21, 2017, at 6:54 AM, Achintha wrote: > > hi michael > > thanks a lot for your quick respond > here i attached the logs of both freeswitch servers and params files. > > testing scenario is first i put a call to fs_master and then crashed it. Then the call landed to fs_slave but RTP did not function. > > i attatched following files and also put it on pastebin > > https://pastebin.freeswitch.org/view/c29ac12a > > On Tue, Jun 20, 2017 at 9:14 PM, Michael Jerris > wrote: > would need a bug report on this one with full logs and config and how to reproduce to look into it. > >> On Jun 20, 2017, at 11:05 AM, Achintha > wrote: >> >> hi all, >> >> I configured two freeswitch servers (freeswitch 1.6.18 on debian 8 ) with call recovery feature. It is working properly on extension to extension , out bound and IVR Calls. >> Then i tried to land an incoming call, the call gets connected to the queue and then freeswitch generate a call to agent and bridge it with queued call and both sides can hear properly. >> Then i crashed the primary server, call got landed to the second freeswitch server but rtp is not functioning, and in the second freeswitch console, it printed "switch_core_sqldb.c:2987 invalied cdr data, call not recovered". But the call was not disconnected. >> >> I used the following configurations >> >> switch.conf.xml : >> >> switch name is same on both servers >> core-db-dsn and core-recovery-db-dsn configured with pgsql >> both sip profiles: >> odbc-dsn configured with pgsql >> track-calls elabled >> >> please provide me a solution to sort out this. >> >> -- >> Best Regards.. >> Achintha Udukumbura >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -- > Best Regards.. > Achintha Udukumbura > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From krice at freeswitch.org Wed Jun 21 15:49:30 2017 From: krice at freeswitch.org (Ken Rice) Date: Wed, 21 Jun 2017 10:49:30 -0500 Subject: [Freeswitch-users] Calling on the community for Bug Marshals In-Reply-To: References: Message-ID: hipchat.freeswitch.org This uses the same user database as Jira etc, but just remember to use the email address associated with your Jira account when you login. From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Joel Serrano Sent: Wednesday, June 21, 2017 10:14 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Calling on the community for Bug Marshals What is the server name for FS's Hipchat? On Wed, Jun 21, 2017 at 5:45 AM, Ítalo Rossi > wrote: Cool! Can you guys join our hipchat ? There's a channel there called Bughunt, join there and we can chat about. On Tue, Jun 20, 2017 at 1:35 PM, > wrote: I would be happy to help. Let me know what the "requirements" are. Sent from my iPhone On Jun 20, 2017, at 10:30 AM, David Villasmil > wrote: Anything i can help with? On Tue, Jun 20, 2017 at 2:46 PM Ítalo Rossi > wrote: This is a great opportunity to learn and to be an expert in FreeSWITCH. This was how I learn a lot! :-) On Mon, Jun 5, 2017 at 4:39 PM, Brian West > wrote: FreeSWITCHers, We are in need of a few good bug marshals, We are trying to get 1.8 ready and out the door and the more help we have testing and working thru patches on JIRA the quicker it will arrive. If you're interested in helping us out email me directly. We are also considering bringing back a few days a week we are sitting in 888 and helping the community out with issues pending in JIRA. Also we are only about 2600 short on the gofund me for the Allison prompts, which will be delivered sometime this week. ;) So help us get over that last little bit this week. Thanks, -- Brian West brian at freeswitch.org Twitter: @FreeSWITCH , @briankwest http://www.freeswitchbook.com http://www.freeswitchcookbook.com Book a phone call (CST) Allison prompts for FreeSWITCH: https://www.gofundme.com/allison-prompts-for-freeswitch Got Bugs? Report them here! | Reddit: /r/freeswitch T:+19184209001 | F:+19184209002 | M:+1918424WEST (9378) Skype:briankwest _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Ítalo Rossi italo at freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at 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 at 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 at 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 at 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 at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Ítalo Rossi italo at freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Wed Jun 21 15:56:36 2017 From: mike at jerris.com (Michael Jerris) Date: Wed, 21 Jun 2017 11:56:36 -0400 Subject: [Freeswitch-users] Non-commercial nature of this mailing list. Message-ID: <88ED8755-E260-4E20-A7E4-1AB454682B87@jerris.com> A reminder that this mailing list is intended for the community to be able to discuss FreeSWITCH related topics, but it IS NOT here for you to advertise or promote your products, recruit for jobs, or other such commercial purposes. List messages with these topics or even signatures that promote your products will trigger messages being rejected or silently discarded. I know i have personally attempted to address this with multiple members of this list who persist in posting things with signatures that advertise their products. Please remove any signatures like that from your posts. Repeat offenders who do not follow this rule may be removed from the list without warning. Also, to those with large privacy signatures on your mailing list posts, it should go without saying that this is a public mailing list, and anything posted here will be on the internet forever. We will not remove any messages from the mailing list archives regardless of the reason (yes it has been requested on multiple occasions in the past). Please be careful not to post material you may regret in the future. Thank you for your understanding to keep this the highest quality content for the community. Mike From agubbe at gmail.com Wed Jun 21 16:28:25 2017 From: agubbe at gmail.com (=?UTF-8?Q?Agust=C3=AD_Ubalde_Bellot?=) Date: Wed, 21 Jun 2017 18:28:25 +0200 Subject: [Freeswitch-users] Freeswitch sslv3 support In-Reply-To: References: Message-ID: Hi Michael, The issue has been closed without any patch attached. Will any solution be implemented? Thaks, Agustí 2017-06-21 9:30 GMT+02:00 Agustí Ubalde Bellot : > Hi Michael, > > I've created this ISSUE: https://freeswitch.org/jira/browse/FS-10406. > > > Regards, > Agustí > > 2017-06-20 17:55 GMT+02:00 Agustí Ubalde Bellot : > >> Hi Michael, >> >> Yes, the version I am using is a development version (1.5.14). In any >> case, I have performed the same tests in version 1.6 and have the same >> behavior. >> Instead, the verto module does block the sslv3 protocol. >> >> >> Thanks, >> Agustí >> >> 2017-06-20 10:45 GMT+02:00 Agustí Ubalde Bellot : >> >>> Hi Michael, >>> >>> I have performed several connection tests forcing the sslv3 protocol >>> over secure web sockets and the connection is established. Instead, the >>> same test connecting to the TLS listening port, the connection is not set. >>> The protocol is successfully disabled in the configuration. >>> The version of FreeSWITCH I'm testing is 1.5.14. Is there any way to >>> prove that the sslv3 protocol is actually disabled in this release? >>> >>> >>> Thanks, >>> Agustí >>> >>> 2017-06-15 10:07 GMT+02:00 Agustí Ubalde Bellot : >>> >>>> Hi Brian, >>>> >>>> Is possible to disable for web socket secure connections too? >>>> >>>> >>>> Thanks, >>>> Agustí >>>> >>>> 2017-06-13 13:24 GMT+02:00 Agustí Ubalde Bellot : >>>> >>>>> Hi all, >>>>> >>>>> Is there a FreeSWITCH update where sslv3 support is disabled? >>>>> >>>>> >>>>> Thanks, >>>>> Agustí >>>>> >>>> >>>> >>> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Wed Jun 21 16:35:30 2017 From: mike at jerris.com (Michael Jerris) Date: Wed, 21 Jun 2017 12:35:30 -0400 Subject: [Freeswitch-users] Freeswitch sslv3 support In-Reply-To: References: Message-ID: <4486858C-7330-4B7D-B95B-F2775F053EE7@jerris.com> Brian pushed fix to tree this morning commit dcc0bf72ec77042acf38172af262a4a2f35c1f48 Author: Brian West Date: Wed Jun 21 08:51:50 2017 -0500 FS-10406: [mod_sofia] mod_sofia secure websocket connections SSLv3 and tls v1.0 is still not disabled #resolve > On Jun 21, 2017, at 12:28 PM, Agustí Ubalde Bellot wrote: > > Hi Michael, > > The issue has been closed without any patch attached. Will any solution be implemented? > > 2017-06-21 9:30 GMT+02:00 Agustí Ubalde Bellot >: > Hi Michael, > > I've created this ISSUE: https://freeswitch.org/jira/browse/FS-10406 . > > 2017-06-20 17:55 GMT+02:00 Agustí Ubalde Bellot >: > Hi Michael, > > Yes, the version I am using is a development version (1.5.14). In any case, I have performed the same tests in version 1.6 and have the same behavior. > Instead, the verto module does block the sslv3 protocol. > > 2017-06-20 10:45 GMT+02:00 Agustí Ubalde Bellot >: > Hi Michael, > > I have performed several connection tests forcing the sslv3 protocol over secure web sockets and the connection is established. Instead, the same test connecting to the TLS listening port, the connection is not set. The protocol is successfully disabled in the configuration. > The version of FreeSWITCH I'm testing is 1.5.14. Is there any way to prove that the sslv3 protocol is actually disabled in this release? > > > 2017-06-15 10:07 GMT+02:00 Agustí Ubalde Bellot >: > Hi Brian, > > Is possible to disable for web socket secure connections too? > > > 2017-06-13 13:24 GMT+02:00 Agustí Ubalde Bellot >: > Hi all, > > Is there a FreeSWITCH update where sslv3 support is disabled? -------------- next part -------------- An HTML attachment was scrubbed... URL: From eschmidbauer at gmail.com Wed Jun 21 19:02:00 2017 From: eschmidbauer at gmail.com (E. Schmidbauer) Date: Wed, 21 Jun 2017 15:02:00 -0400 Subject: [Freeswitch-users] IVR menu-back after a timeout In-Reply-To: References: Message-ID: Not sure if it's documented but there is exec-on-max-timeouts And exec-on-max-failures You can use those to go to another ivr (top or back) On Jun 21, 2017 3:48 AM, "Giovanni Maruzzelli" wrote: > On 21 June 2017 at 00:55, Steven Schoch > wrote: > >> Is it possible for an IVR (in XML) to do a "menu-back" to go back to the >> top-level menu after a timeout? >> >> > nope > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian at freeswitch.org Wed Jun 21 19:10:54 2017 From: brian at freeswitch.org (Brian West) Date: Wed, 21 Jun 2017 14:10:54 -0500 Subject: [Freeswitch-users] Allison Prompts Message-ID: FreeSWITCHers, We accepted delivery of the sound set, they are now in GIT, We'll get those packaged up soon, Please help us meet our goals! https://www.gofundme.com/allison-prompts-for-freeswitch https://freeswitch.org/stash/projects/FS/repos/freeswitch-sounds/browse/en/us/allison Donate TODAY! Thanks, -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Book a phone call (CST) Allison prompts for FreeSWITCH: *https://www.gofundme.com/allison-prompts-for-freeswitch* Got Bugs? Report them here ! | Reddit: /r/freeswitch *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: From thesipguy at gmail.com Wed Jun 21 19:22:28 2017 From: thesipguy at gmail.com (Schneur Rosenberg) Date: Wed, 21 Jun 2017 22:22:28 +0300 Subject: [Freeswitch-users] IVR menu-back after a timeout In-Reply-To: References: Message-ID: E. Schmidbauer thank you very much, I asked so many people about this feature and I was told its not possible, I will test it later. On Wed, Jun 21, 2017 at 10:02 PM, E. Schmidbauer wrote: > Not sure if it's documented but there is > exec-on-max-timeouts > > And > > exec-on-max-failures > > You can use those to go to another ivr (top or back) > > On Jun 21, 2017 3:48 AM, "Giovanni Maruzzelli" wrote: > >> On 21 June 2017 at 00:55, Steven Schoch > > wrote: >> >>> Is it possible for an IVR (in XML) to do a "menu-back" to go back to the >>> top-level menu after a timeout? >>> >>> >> nope >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From gmaruzz at gmail.com Wed Jun 21 19:24:18 2017 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Wed, 21 Jun 2017 21:24:18 +0200 Subject: [Freeswitch-users] Allison Prompts In-Reply-To: References: Message-ID: I, for one, welcome our new overlady ! Thee yet donated have not, donate already ! -giovanni On 21 June 2017 at 21:10, Brian West wrote: > FreeSWITCHers, > > We accepted delivery of the sound set, they are now in GIT, We'll get > those packaged up soon, Please help us meet our goals! > > https://www.gofundme.com/allison-prompts-for-freeswitch > > https://freeswitch.org/stash/projects/FS/repos/freeswitch- > sounds/browse/en/us/allison > > Donate TODAY! > > > Thanks, > > > -- > > *Brian West* > brian at freeswitch.org > > *Twitter: @FreeSWITCH , @briankwest* > > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > Book a phone call (CST) > > Allison prompts for FreeSWITCH: > > *https://www.gofundme.com/allison-prompts-for-freeswitch* > > > Got Bugs? Report them here ! | Reddit: > /r/freeswitch > > *T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002 <(918)%20420-9002> > | *M:*+1918424WEST (9378) > *Skype:*briankwest > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at 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 OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: From ksrigo at gmail.com Wed Jun 21 19:26:11 2017 From: ksrigo at gmail.com (Srigo Kana) Date: Wed, 21 Jun 2017 21:26:11 +0200 Subject: [Freeswitch-users] Routing Calls to registered endpoints. In-Reply-To: References: Message-ID: In your dialplan (30_routing.xml) set sip_redirect_context=to_anything. Then when your Jerasoft will send you back the redirect you will be able to catch the call in the context you defined above and do the bridge using sip_redirct_contact_0 to your internal profile to find the local user. Srigo Sent from my iPhone > On 20 Jun 2017, at 21:01, Jospeh Waite wrote: > > Hi Guys. > > Final question of the day. I promise!! > > I have 2 sofia profiles, one for IP authenticated calls and one for Sip Registrations. There both on port 5060 on 2 different IP addresses. > This is basically a Wholesale/retail billing/switching platform. We have a JeraSoft VCS radius based billing and Sip Redirect server. > > Everything works ok, except for inbound calls from DID provider destined for a registered extension. > > Call comes in on the IP profile, which then sends a request to the JeraSoft SIP Redirect server, which replies with a 300 multiple choices as follows. > > SIP/2.0 300 Multiple Choices > Via: SIP/2.0/UDP {ip of FS IP profile};rport;branch=z9hG4bKSKFBFy6K81U9H > From: "07966677711" ;tag=S6Xmmj7meS2Fr > To: > Contact: ;q=1.00 > Call-ID: aed22c59-d082-1235-baa2-363165383663 > CSeq: 108654212 INVITE > Max-Forwards: 67 > Content-Length: 0 > Server: JeraSoft VCS SIP Redirect Server > > joehouse is the username of a registered user, registered to the reg profile. > > Now FS instead of connecting the call to the registered user it sends a new request to the Jerasoft redirect. > > How would I get it to route the call to the registered user? > > public/30_routing.xml > > > > > > > > > > > > > > > > > > --> > > > > > > > > public.xml > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From gmaruzz at gmail.com Wed Jun 21 19:30:20 2017 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Wed, 21 Jun 2017 21:30:20 +0200 Subject: [Freeswitch-users] IVR menu-back after a timeout In-Reply-To: References: Message-ID: https://freeswitch.org/confluence/display/FREESWITCH/mod_ dptools:+IVR+Menu#mod_dptools:IVRMenu-Options This is what you want to exec if/when the max timeouts (the number of times the menu has been played without input from caller) has been reached. Not sure is what the OP was looking for, but maybe he can use it... -giovanni On 21 June 2017 at 21:22, Schneur Rosenberg wrote: > E. Schmidbauer thank you very much, I asked so many people about this > feature and I was told its not possible, I will test it later. > > On Wed, Jun 21, 2017 at 10:02 PM, E. Schmidbauer > wrote: > >> Not sure if it's documented but there is >> exec-on-max-timeouts >> >> And >> >> exec-on-max-failures >> >> You can use those to go to another ivr (top or back) >> >> On Jun 21, 2017 3:48 AM, "Giovanni Maruzzelli" wrote: >> >>> On 21 June 2017 at 00:55, Steven Schoch >> com> wrote: >>> >>>> Is it possible for an IVR (in XML) to do a "menu-back" to go back to >>>> the top-level menu after a timeout? >>>> >>>> >>> nope >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at 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 at 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 at 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 at 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 at 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 at 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 OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: From jungleboogie0 at gmail.com Thu Jun 22 00:56:50 2017 From: jungleboogie0 at gmail.com (jungle Boogie) Date: Wed, 21 Jun 2017 17:56:50 -0700 Subject: [Freeswitch-users] switch_core_sqldb.c:2987 invalied cdr data, call not recoverd In-Reply-To: References: <5A5DBC29-DCE6-417F-8B48-2BE9A3390523@jerris.com> Message-ID: Did you create a jira? -------------- next part -------------- An HTML attachment was scrubbed... URL: From andrew.keil at visytel.com Thu Jun 22 01:24:00 2017 From: andrew.keil at visytel.com (Andrew Keil) Date: Thu, 22 Jun 2017 01:24:00 +0000 Subject: [Freeswitch-users] Re- PocketSphinx Message-ID: To FreeSWITCH Users, My question is a simple one. Is there anyone that has recently used and got working PocketSphinx inside the latest production build of FreeSWITCH on Linux (CentOS 6 or 7)? I have currently managed to get it working to an extent, however the results are very mixed (eg. sometimes even the simple pizza_yesno grammar always returns yes at 100% confidence when I say "no"). I know it is "Free", however I am hoping that someone has a good grasp on how it works to enable me to improve the recognition and make it work more successfully. It could well be something I have done (or not done) inside my setup. My aim would be to update the confluence documentation to enable this information to be public to assist anyone else wishing to use mod_pocketsphinx. Cheers, Andrew Keil -------------- next part -------------- An HTML attachment was scrubbed... URL: From s.safarov at gmail.com Thu Jun 22 04:25:23 2017 From: s.safarov at gmail.com (Sergey Safarov) Date: Thu, 22 Jun 2017 04:25:23 +0000 Subject: [Freeswitch-users] Re- PocketSphinx In-Reply-To: References: Message-ID: I have done this for one project. I cannot provide source as example. Customer disallow do it. Sergey чт, 22 июн. 2017 г. в 4:27, Andrew Keil : > To FreeSWITCH Users, > > > > My question is a simple one. Is there anyone that has recently used and > got working PocketSphinx inside the latest production build of FreeSWITCH > on Linux (CentOS 6 or 7)? > > > > I have currently managed to get it working to an extent, however the > results are very mixed (eg. sometimes even the simple pizza_yesno grammar > always returns yes at 100% confidence when I say “no”). > > > > I know it is “Free”, however I am hoping that someone has a good grasp on > how it works to enable me to improve the recognition and make it work more > successfully. It could well be something I have done (or not done) inside > my setup. > > > > My aim would be to update the confluence documentation to enable this > information to be public to assist anyone else wishing to use > mod_pocketsphinx. > > > > Cheers, > > > > Andrew Keil > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From achinthau at gmail.com Thu Jun 22 06:29:51 2017 From: achinthau at gmail.com (Achintha) Date: Thu, 22 Jun 2017 11:59:51 +0530 Subject: [Freeswitch-users] switch_core_sqldb.c:2987 invalied cdr data, call not recoverd In-Reply-To: References: <5A5DBC29-DCE6-417F-8B48-2BE9A3390523@jerris.com> Message-ID: hi all, I created a bug on jira https://freeswitch.org/jira/browse/FS-10418 On Thu, Jun 22, 2017 at 6:26 AM, jungle Boogie wrote: > Did you create a jira? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Best Regards.. Achintha Udukumbura -------------- next part -------------- An HTML attachment was scrubbed... URL: From findmeinwland at gmail.com Thu Jun 22 10:18:56 2017 From: findmeinwland at gmail.com (Artur Mega) Date: Thu, 22 Jun 2017 15:18:56 +0500 Subject: [Freeswitch-users] Re- PocketSphinx In-Reply-To: References: Message-ID: I did some tests on previous company where I worked. I used russian language model. Recognizing was good. I don't have source code, but I can advice to compile PS from source, and not the latest version. I don't remeber the version I used. It was on Debian. -------------- next part -------------- An HTML attachment was scrubbed... URL: From alexandr.popov at iqoption.com Thu Jun 22 09:46:10 2017 From: alexandr.popov at iqoption.com (Alexandr Popov) Date: Thu, 22 Jun 2017 12:46:10 +0300 Subject: [Freeswitch-users] Re- PocketSphinx In-Reply-To: References: Message-ID: i had played around and got 75%-95%accuracy(depends of channel quality) with russian lang model. It would be great if this module could work with binary grammar instead of jsgf but anyway its good enough. 2017-06-22 7:25 GMT+03:00 Sergey Safarov : > I have done this for one project. > I cannot provide source as example. Customer disallow do it. > > Sergey > > чт, 22 июн. 2017 г. в 4:27, Andrew Keil : > >> To FreeSWITCH Users, >> >> >> >> My question is a simple one. Is there anyone that has recently used and >> got working PocketSphinx inside the latest production build of FreeSWITCH >> on Linux (CentOS 6 or 7)? >> >> >> >> I have currently managed to get it working to an extent, however the >> results are very mixed (eg. sometimes even the simple pizza_yesno grammar >> always returns yes at 100% confidence when I say “no”). >> >> >> >> I know it is “Free”, however I am hoping that someone has a good grasp on >> how it works to enable me to improve the recognition and make it work more >> successfully. It could well be something I have done (or not done) inside >> my setup. >> >> >> >> My aim would be to update the confluence documentation to enable this >> information to be public to assist anyone else wishing to use >> mod_pocketsphinx. >> >> >> >> Cheers, >> >> >> >> Andrew Keil >> >> >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From afarooqa at gmail.com Thu Jun 22 13:49:33 2017 From: afarooqa at gmail.com (Faruq Ahmad) Date: Thu, 22 Jun 2017 18:49:33 +0500 Subject: [Freeswitch-users] Javascript FileIO Message-ID: Hi, I have a FS dialplan that fetches callflow from a JSON file. File is read using var JsonFd = new FileIO(JsonPath, 'r'); JsonFd.read(2048); result = JsonFd.data(); var menu = JSON.parse(result), Is there anyway I can read the whole file in one attempt, i.e. when a smaller size was given to the .read() function it wouldn't read the complete file. I have increased the value of size for read function however my concern is in the long run file size might increase and parse would get an incomplete JSON. Is there anyway I can get filesize from FileIO object or detect EOF from the read buffer to make sure the whole file is loaded? Also if I increase the size for read fucntion buffer way over the estimated filesizes, is there any guarantee that no garbage values will be read from the disk after the EOF? -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Thu Jun 22 15:07:31 2017 From: mike at jerris.com (Michael Jerris) Date: Thu, 22 Jun 2017 11:07:31 -0400 Subject: [Freeswitch-users] Re- PocketSphinx In-Reply-To: References: Message-ID: <3B979CA2-9564-4EEF-A315-B5CF59BF89A1@jerris.com> it has troubles with some codecs possibly… > On Jun 21, 2017, at 9:24 PM, Andrew Keil wrote: > > To FreeSWITCH Users, > > My question is a simple one. Is there anyone that has recently used and got working PocketSphinx inside the latest production build of FreeSWITCH on Linux (CentOS 6 or 7)? > > I have currently managed to get it working to an extent, however the results are very mixed (eg. sometimes even the simple pizza_yesno grammar always returns yes at 100% confidence when I say “no”). > > I know it is “Free”, however I am hoping that someone has a good grasp on how it works to enable me to improve the recognition and make it work more successfully. It could well be something I have done (or not done) inside my setup. > > My aim would be to update the confluence documentation to enable this information to be public to assist anyone else wishing to use mod_pocketsphinx. > > Cheers, > > Andrew Keil > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From nabeel.10.ahmed at gmail.com Thu Jun 22 20:24:26 2017 From: nabeel.10.ahmed at gmail.com (Nabeel Ahmad) Date: Fri, 23 Jun 2017 01:24:26 +0500 Subject: [Freeswitch-users] Call limit on RTP-IP Address In-Reply-To: References: <9cad01d2e9ce$3ab34e50$b019eaf0$@freeswitch.org> <5a52f788-d3b1-a3fb-4e0d-daeb247eb782@nevian.org> <9d7b01d2e9d3$366f97e0$a34ec7a0$@freeswitch.org> <9e2901d2e9dd$391c0eb0$ab542c10$@freeswitch.org> Message-ID: Hi, Ok call limit per media ip can be achieved in many ways though i was looking some straight FSish method . can we use rtp over tcp ? I know its a bad idea but where extreme udp filtering is done , only way is to work on tcp. So can FS work with coturn to relay media in tcp some how (or any other way )? If have to pay for this will do. On Wed, Jun 21, 2017 at 3:01 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > Its not possible. Maybe file a bounty. > > On Tue, Jun 20, 2017 at 2:44 PM Nabeel Ahmad > wrote: > >> Umm...I am on limit page since Morning . limit on any user , outbound >> gateway all is working well. they are all in dialplan . How can i limit >> inbound profile this i still can't figure from the link given. >> >> And one question for rtp-ip value as i was doing it some other way . >> If i do like you have mentioned ,sofia replies with this sdp . >> >> ** >> >> o=FreeSWITCH 1497955081 1497955082 IN IP4 192.168.1.1,192.168.1.2,192. >> 168.1.4 >> s=FreeSWITCH >> *c=IN IP4 192.168.1.1,192.168.1.2,192.168.1.4 * >> >> I am not sure if this is valid sdp attribute or not. >> >> I am doing like this >> >> >> >> >> Which on each call offers one ip in sdp like this . >> o=FreeSWITCH 1497961414 1497961415 IN IP4 192.168.1.3 >> s=FreeSWITCH >> c=IN IP4 192.168.1.3 >> >> and this on another call . >> >> o=FreeSWITCH 1497961414 1497961415 IN IP4 192.168.1.1 >> s=FreeSWITCH >> c=IN IP4 192.168.1.1 >> >> >> >> On Tue, Jun 20, 2017 at 8:52 PM, Ken Rice wrote: >> >>> No Defining multiple RTP IPs has been there for a while.... you don’t >>> define them in different lines, the parser will filter out previous ones, >>> you define them all together >>> >>> >>> >>> The stack will then round robin them. This feature was added several >>> years ago so that FreeSWITCH can handle the required RTP load in traffic >>> flows that can exceed that of 1Gig-E network connections while only using 1 >>> SIP Profile for traffic. It's still useful although less of a requirement >>> with 10GigE coming down in price. (you can now find managed 48port 10GE >>> switches, NICs and Cables on the secondary market for a combined cost under >>> $200/port now. >>> >>> >>> -----Original Message----- >>> From: FreeSWITCH-users [mailto:freeswitch-users- >>> bounces at lists.freeswitch.org] On Behalf Of Serge S. Yuriev >>> Sent: Tuesday, June 20, 2017 10:35 AM >>> To: freeswitch-users at lists.freeswitch.org >>> Subject: Re: [Freeswitch-users] Call limit on RTP-IP Address >>> >>> I mean if we can define multiple IPs at all. I was under impression >>> parser uses only one last defined. >>> >>> On 20/06/17 17:40, Ken Rice wrote: >>> > This is not a bug... there is just no way to select which IP or port >>> is used nor is there an effective way to limit it short of just killing the >>> call due to the way the IP/Port allocation works in the RTP stack currently. >>> > >>> > -----Original Message----- >>> > From: FreeSWITCH-users >>> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of >>> > Serge S. Yuriev >>> > Sent: Tuesday, June 20, 2017 9:17 AM >>> > To: freeswitch-users at lists.freeswitch.org >>> > Subject: Re: [Freeswitch-users] Call limit on RTP-IP Address >>> > >>> > Is this multi-IP config expected to work this way or it's >>> eventuality/bug? >>> > >>> > On 20/06/17 17:05, Ken Rice wrote: >>> >> Theres no way to limit or select which IP/Port combination is used >>> >> from the available RTP IP/Port Range in the config you have. The >>> >> only way to do this would be to create a profile for each IP and then >>> >> limit the number of calls per profile >>> >> >>> >> *From:* FreeSWITCH-users >>> >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of >>> >> *Nabeel Ahmad >>> >> *Sent:* Tuesday, June 20, 2017 4:16 AM >>> >> *To:* freeswitch-users at lists.freeswitch.org >>> >> *Subject:* [Freeswitch-users] Call limit on RTP-IP Address >>> >> >>> >> Hello all, >>> >> >>> >> If we assign more then one RTP-IP parameter to a sip profile , they >>> >> are used in round robin. >>> >> >>> >> Its perfect . I want to know is there a way to use some limit on that >>> ip ? >>> >> >>> >> Say i've 5 ip address listening on box , and i want one concurrent >>> >> call limit on each media ip. >>> >> >>> >> How can i set limit on profile level or set from dialplan. >>> >> >>> >> I tried to do but it didn't work . >>> >> >>> >> Any help or advice will be highly appreciated >>> >> >>> >> Thanking all >>> >> >>> >> Nabeel. >>> > >>> > >>> > -- >>> > Serge S. Yuriev >>> > Lead VoIP engineer >>> > >>> > >>> > ______________________________________________________________________ >>> > ___ Professional FreeSWITCH Consulting Services: >>> > consulting at 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 at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use >>> > rs >>> > http://www.freeswitch.org >>> > >>> > >>> > ______________________________________________________________________ >>> > ___ Professional FreeSWITCH Consulting Services: >>> > consulting at 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 at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-use >>> > rs >>> > http://www.freeswitch.org >>> > >>> >>> -- >>> Serge S. Yuriev >>> Lead VoIP engineer >>> >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at 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 at 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 at 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 at 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 at 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 at 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 ♬ @anthmfs ♬ @FreeSWITCH ♬ > > ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ > http://twitter.com/FreeSWITCH > ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ > * > > ClueCon Weekly Development Call > ☎ sip:888 at conference.freeswitch.org ☎ +19193869900 <(919)%20386-9900> > > https://www.youtube.com/watch?v=9XXgW34t40s > https://www.youtube.com/watch?v=NLaDpGQuZDA > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From anthony.minessale at gmail.com Thu Jun 22 20:55:52 2017 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 22 Jun 2017 15:55:52 -0500 Subject: [Freeswitch-users] Call limit on RTP-IP Address In-Reply-To: References: <9cad01d2e9ce$3ab34e50$b019eaf0$@freeswitch.org> <5a52f788-d3b1-a3fb-4e0d-daeb247eb782@nevian.org> <9d7b01d2e9d3$366f97e0$a34ec7a0$@freeswitch.org> <9e2901d2e9dd$391c0eb0$ab542c10$@freeswitch.org> Message-ID: Its possible in theory as there is a spec for it but it would require both ends supporting it and would have to be researched and a price figured out. You can contact consulting at freeswitch.org if you would like. On Thu, Jun 22, 2017 at 3:24 PM, Nabeel Ahmad wrote: > Hi, > Ok call limit per media ip can be achieved in many ways though i was > looking some straight FSish method . > can we use rtp over tcp ? I know its a bad idea but where extreme udp > filtering is done , only way is to work on tcp. So can FS work with coturn > to relay media in tcp some how (or any other way )? If have to pay for this > will do. > > > On Wed, Jun 21, 2017 at 3:01 AM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> Its not possible. Maybe file a bounty. >> >> On Tue, Jun 20, 2017 at 2:44 PM Nabeel Ahmad >> wrote: >> >>> Umm...I am on limit page since Morning . limit on any user , outbound >>> gateway all is working well. they are all in dialplan . How can i limit >>> inbound profile this i still can't figure from the link given. >>> >>> And one question for rtp-ip value as i was doing it some other way . >>> If i do like you have mentioned ,sofia replies with this sdp . >>> >>> ** >>> >>> o=FreeSWITCH 1497955081 1497955082 IN IP4 192.168.1.1,192.168.1.2,192.16 >>> 8.1.4 >>> s=FreeSWITCH >>> *c=IN IP4 192.168.1.1,192.168.1.2,192.168.1.4 * >>> >>> I am not sure if this is valid sdp attribute or not. >>> >>> I am doing like this >>> >>> >>> >>> >>> Which on each call offers one ip in sdp like this . >>> o=FreeSWITCH 1497961414 1497961415 IN IP4 192.168.1.3 >>> s=FreeSWITCH >>> c=IN IP4 192.168.1.3 >>> >>> and this on another call . >>> >>> o=FreeSWITCH 1497961414 1497961415 IN IP4 192.168.1.1 >>> s=FreeSWITCH >>> c=IN IP4 192.168.1.1 >>> >>> >>> >>> On Tue, Jun 20, 2017 at 8:52 PM, Ken Rice wrote: >>> >>>> No Defining multiple RTP IPs has been there for a while.... you don’t >>>> define them in different lines, the parser will filter out previous ones, >>>> you define them all together >>>> >>>> >>>> >>>> The stack will then round robin them. This feature was added several >>>> years ago so that FreeSWITCH can handle the required RTP load in traffic >>>> flows that can exceed that of 1Gig-E network connections while only using 1 >>>> SIP Profile for traffic. It's still useful although less of a requirement >>>> with 10GigE coming down in price. (you can now find managed 48port 10GE >>>> switches, NICs and Cables on the secondary market for a combined cost under >>>> $200/port now. >>>> >>>> >>>> -----Original Message----- >>>> From: FreeSWITCH-users [mailto:freeswitch-users-bounc >>>> es at lists.freeswitch.org] On Behalf Of Serge S. Yuriev >>>> Sent: Tuesday, June 20, 2017 10:35 AM >>>> To: freeswitch-users at lists.freeswitch.org >>>> Subject: Re: [Freeswitch-users] Call limit on RTP-IP Address >>>> >>>> I mean if we can define multiple IPs at all. I was under impression >>>> parser uses only one last defined. >>>> >>>> On 20/06/17 17:40, Ken Rice wrote: >>>> > This is not a bug... there is just no way to select which IP or port >>>> is used nor is there an effective way to limit it short of just killing the >>>> call due to the way the IP/Port allocation works in the RTP stack currently. >>>> > >>>> > -----Original Message----- >>>> > From: FreeSWITCH-users >>>> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of >>>> > Serge S. Yuriev >>>> > Sent: Tuesday, June 20, 2017 9:17 AM >>>> > To: freeswitch-users at lists.freeswitch.org >>>> > Subject: Re: [Freeswitch-users] Call limit on RTP-IP Address >>>> > >>>> > Is this multi-IP config expected to work this way or it's >>>> eventuality/bug? >>>> > >>>> > On 20/06/17 17:05, Ken Rice wrote: >>>> >> Theres no way to limit or select which IP/Port combination is used >>>> >> from the available RTP IP/Port Range in the config you have. The >>>> >> only way to do this would be to create a profile for each IP and then >>>> >> limit the number of calls per profile >>>> >> >>>> >> *From:* FreeSWITCH-users >>>> >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of >>>> >> *Nabeel Ahmad >>>> >> *Sent:* Tuesday, June 20, 2017 4:16 AM >>>> >> *To:* freeswitch-users at lists.freeswitch.org >>>> >> *Subject:* [Freeswitch-users] Call limit on RTP-IP Address >>>> >> >>>> >> Hello all, >>>> >> >>>> >> If we assign more then one RTP-IP parameter to a sip profile , they >>>> >> are used in round robin. >>>> >> >>>> >> Its perfect . I want to know is there a way to use some limit on >>>> that ip ? >>>> >> >>>> >> Say i've 5 ip address listening on box , and i want one concurrent >>>> >> call limit on each media ip. >>>> >> >>>> >> How can i set limit on profile level or set from dialplan. >>>> >> >>>> >> I tried to do but it didn't work . >>>> >> >>>> >> Any help or advice will be highly appreciated >>>> >> >>>> >> Thanking all >>>> >> >>>> >> Nabeel. >>>> > >>>> > >>>> > -- >>>> > Serge S. Yuriev >>>> > Lead VoIP engineer >>>> > >>>> > >>>> > ____________________________________________________________ >>>> __________ >>>> > ___ Professional FreeSWITCH Consulting Services: >>>> > consulting at 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 at lists.freeswitch.org >>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ >>>> freeswitch-use >>>> > rs >>>> > http://www.freeswitch.org >>>> > >>>> > >>>> > ____________________________________________________________ >>>> __________ >>>> > ___ Professional FreeSWITCH Consulting Services: >>>> > consulting at 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 at lists.freeswitch.org >>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ >>>> freeswitch-use >>>> > rs >>>> > http://www.freeswitch.org >>>> > >>>> >>>> -- >>>> Serge S. Yuriev >>>> Lead VoIP engineer >>>> >>>> >>>> ____________________________________________________________ >>>> _____________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at 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 at 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 at 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 at 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 at 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 at 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 ♬ @anthmfs ♬ @FreeSWITCH ♬ >> >> ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ >> http://twitter.com/FreeSWITCH >> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ >> * >> >> ClueCon Weekly Development Call >> ☎ sip:888 at conference.freeswitch.org ☎ +19193869900 <(919)%20386-9900> >> >> https://www.youtube.com/watch?v=9XXgW34t40s >> https://www.youtube.com/watch?v=NLaDpGQuZDA >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at 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 at 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 at 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 ♬ @anthmfs ♬ @FreeSWITCH ♬ ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ * ClueCon Weekly Development Call ☎ sip:888 at conference.freeswitch.org ☎ +19193869900 https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA -------------- next part -------------- An HTML attachment was scrubbed... URL: From xxxman2008 at 126.com Fri Jun 23 08:36:18 2017 From: xxxman2008 at 126.com (Raymond) Date: Fri, 23 Jun 2017 16:36:18 +0800 (CST) Subject: [Freeswitch-users] Javascript FileIO In-Reply-To: References: Message-ID: <2e9e2174.97ca.15cd41840cd.Coremail.xxxman2008@126.com> Hi , This will help you . https://wiki.freeswitch.org/wiki/File Raymond At 2017-06-22 21:49:33, "Faruq Ahmad" wrote: Hi, I have a FS dialplan that fetches callflow from a JSON file. File is read using var JsonFd = new FileIO(JsonPath, 'r'); JsonFd.read(2048); result = JsonFd.data(); var menu = JSON.parse(result), Is there anyway I can read the whole file in one attempt, i.e. when a smaller size was given to the .read() function it wouldn't read the complete file. I have increased the value of size for read function however my concern is in the long run file size might increase and parse would get an incomplete JSON. Is there anyway I can get filesize from FileIO object or detect EOF from the read buffer to make sure the whole file is loaded? Also if I increase the size for read fucntion buffer way over the estimated filesizes, is there any guarantee that no garbage values will be read from the disk after the EOF? -------------- next part -------------- An HTML attachment was scrubbed... URL: From gmaruzz at gmail.com Fri Jun 23 08:46:23 2017 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Fri, 23 Jun 2017 10:46:23 +0200 Subject: [Freeswitch-users] SKYPE: mod_skypopen END-OF-LIFE Message-ID: Hello FreeSWITCHers, Microsoft decided to kill all old Skype clients July the 1st, and new clients do not support APIs to interact with calls. So, no way mod_skypopen can continue to work. Has been a nice trip, a lot has changed in the Real Time Communication world since first chan_skypiax, then mod_skypopen bridged Skype and open standard calls. mod_skypopen sez: "GOODBYE WORLD!" -giovanni -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: From mitul at enterux.com Fri Jun 23 09:26:51 2017 From: mitul at enterux.com (Mitul Limbani) Date: Fri, 23 Jun 2017 14:56:51 +0530 Subject: [Freeswitch-users] SKYPE: mod_skypopen END-OF-LIFE In-Reply-To: References: Message-ID: Sad to see this die :( Mitul Regards, Mitul Limbani, Business Head, Enterux Solutions Pvt. Ltd. 110 Reena Complex, Opp. Nathani Steel, Vidyavihar (W), Mumbai - 400 086. India http://www.enterux.com/ http://www.entvoice.com/ email: mitul at enterux.in DID: +91-22-71967196 Cell: +91-9820332422 On Fri, Jun 23, 2017 at 2:16 PM, Giovanni Maruzzelli wrote: > Hello FreeSWITCHers, > > > Microsoft decided to kill all old Skype clients July the 1st, and new > clients do not support APIs to interact with calls. > > So, no way mod_skypopen can continue to work. > > Has been a nice trip, a lot has changed in the Real Time Communication > world since first chan_skypiax, then mod_skypopen bridged Skype and open > standard calls. > > mod_skypopen sez: "GOODBYE WORLD!" > > > -giovanni > > > > -- > > Sincerely, > > Giovanni Maruzzelli > OpenTelecom.IT > cell: +39 347 266 56 18 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From gmaruzz at gmail.com Fri Jun 23 09:29:54 2017 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Fri, 23 Jun 2017 11:29:54 +0200 Subject: [Freeswitch-users] SKYPE: mod_skypopen END-OF-LIFE In-Reply-To: References: Message-ID: Ciao Mitul !!!! A friendly and dear hug ! -giovanni On 23 June 2017 at 11:26, Mitul Limbani wrote: > Sad to see this die :( > > Mitul > > Regards, > Mitul Limbani, > Business Head, > Enterux Solutions Pvt. Ltd. > 110 Reena Complex, Opp. Nathani Steel, > Vidyavihar (W), Mumbai - 400 086. India > http://www.enterux.com/ > http://www.entvoice.com/ > email: mitul at enterux.in > DID: +91-22-71967196 <+91%2022%207196%207196> > Cell: +91-9820332422 <+91%2098203%2032422> > > On Fri, Jun 23, 2017 at 2:16 PM, Giovanni Maruzzelli > wrote: > >> Hello FreeSWITCHers, >> >> >> Microsoft decided to kill all old Skype clients July the 1st, and new >> clients do not support APIs to interact with calls. >> >> So, no way mod_skypopen can continue to work. >> >> Has been a nice trip, a lot has changed in the Real Time Communication >> world since first chan_skypiax, then mod_skypopen bridged Skype and open >> standard calls. >> >> mod_skypopen sez: "GOODBYE WORLD!" >> >> >> -giovanni >> >> >> >> -- >> >> Sincerely, >> >> Giovanni Maruzzelli >> OpenTelecom.IT >> cell: +39 347 266 56 18 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at 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 at 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 at 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 OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: From kd.dhamecha at gmail.com Fri Jun 23 10:54:36 2017 From: kd.dhamecha at gmail.com (Kashyap Dhamecha) Date: Fri, 23 Jun 2017 16:24:36 +0530 Subject: [Freeswitch-users] POST additional channel variable in xml_curl - fs_curl Message-ID: Hello All, I have configured mod_xml_curl and did binding of the directory and configuration. To generate XML file I am using fs_curl and everything is working fine. Recently I was testing mod_directory module and found xml_curl will be used to lookup for names. When lookup event is called it is passing some predefined elements which can be accessed to generate XML file. Along with this predefined elements is there any way to pass any channel variable in this lookup and that variable can be used to add some logic in generating XML file. Initially I found "enable-post-var" param in xm_curl.conf.xml will do it. But it did not work. So is it possible to send any channel variable to access in this lookup? Currently, I am able to access following elements in fs_curl. [hostname] => Backend [section] => directory [tag_name] => domain [key_name] => name [key_value] => 192.168.3.47 [Event-Name] => REQUEST_PARAMS [Core-UUID] => a6470f48-ea06-409e-a4cb-1678e0c9a0d6 [FreeSWITCH-Hostname] => Backend [FreeSWITCH-Switchname] => Backend [FreeSWITCH-IPv4] => 192.168.3.47 [FreeSWITCH-IPv6] => ::1 [Event-Date-Local] => 2017-06-23 15:39:43 [Event-Date-GMT] => Fri, 23 Jun 2017 10:09:43 GMT [Event-Date-Timestamp] => 1498212583180796 [Event-Calling-File] => mod_directory.c [Event-Calling-Function] => populate_database [Event-Calling-Line-Number] => 633 [Event-Sequence] => 132072 Thank you very much in advance. -- With kind regards.... Kashyap Dhamecha -------------- next part -------------- An HTML attachment was scrubbed... URL: From yu at yu-boot.ru Fri Jun 23 11:52:11 2017 From: yu at yu-boot.ru (Yu Boot) Date: Fri, 23 Jun 2017 14:52:11 +0300 Subject: [Freeswitch-users] Include text file in XML dialplan? In-Reply-To: References: Message-ID: <97617996-90a5-9a30-0497-67e06d941e27@yu-boot.ru> Is it possible to just include arbitrary text file to some place in XML dialplan? I want to add or remove some extension and condition entries automatically, without parsing and building entire dialplan XML. From luca.pradovera at gmail.com Fri Jun 23 12:39:23 2017 From: luca.pradovera at gmail.com (Luca Pradovera) Date: Fri, 23 Jun 2017 14:39:23 +0200 Subject: [Freeswitch-users] Verto specification Message-ID: Hello, one of our clients has a requirement to use Verto in a mobile application (outside of a WebView). That means the JQuery dependency can't be met, and we need to figure out a way around it. I as thinking of just reimplementing the signaling plane without any JQuery dependency and taking it from there. Is the actual JSON protocol documented anywhere? I can also do a Verto Communicator install and sniff traffic, but I was looking if there was anything more structured. Alternatively, does anyone have better ideas to offer? Thanks! Best regards, Luca Pradovera -------------- next part -------------- An HTML attachment was scrubbed... URL: From gmaruzz at gmail.com Fri Jun 23 12:40:10 2017 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Fri, 23 Jun 2017 14:40:10 +0200 Subject: [Freeswitch-users] Include text file in XML dialplan? In-Reply-To: <97617996-90a5-9a30-0497-67e06d941e27@yu-boot.ru> References: <97617996-90a5-9a30-0497-67e06d941e27@yu-boot.ru> Message-ID: all XML dialplan and configs are based on including files. Obviously the included files must be XML... On 23 June 2017 at 13:52, Yu Boot wrote: > Is it possible to just include arbitrary text file to some place in XML > dialplan? I want to add or remove some extension and condition entries > automatically, without parsing and building entire dialplan XML. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at 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 OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: From lexxua at gmail.com Fri Jun 23 13:07:32 2017 From: lexxua at gmail.com (Volodymyr Fedorov) Date: Fri, 23 Jun 2017 15:07:32 +0200 Subject: [Freeswitch-users] SKYPE: mod_skypopen END-OF-LIFE In-Reply-To: References: Message-ID: Thanks for your work Giovanni, But I have a question. Did somebody look to ability to bridge webrtc Skype client and Freeswitch ? Current version Skype for linux is "just" chromium page is`t it. For chat protocol exist implementation for Web-skype: https://github.com/EionRobb/skype4pidgin/tree/master/skypeweb Thanks. On Fri, Jun 23, 2017 at 11:29 AM, Giovanni Maruzzelli wrote: > Ciao Mitul !!!! > > A friendly and dear hug ! > -giovanni > > > > On 23 June 2017 at 11:26, Mitul Limbani wrote: > >> Sad to see this die :( >> >> Mitul >> >> Regards, >> Mitul Limbani, >> Business Head, >> Enterux Solutions Pvt. Ltd. >> 110 Reena Complex, Opp. Nathani Steel, >> Vidyavihar (W), Mumbai - 400 086. India >> http://www.enterux.com/ >> http://www.entvoice.com/ >> email: mitul at enterux.in >> DID: +91-22-71967196 <+91%2022%207196%207196> >> Cell: +91-9820332422 <+91%2098203%2032422> >> >> On Fri, Jun 23, 2017 at 2:16 PM, Giovanni Maruzzelli >> wrote: >> >>> Hello FreeSWITCHers, >>> >>> >>> Microsoft decided to kill all old Skype clients July the 1st, and new >>> clients do not support APIs to interact with calls. >>> >>> So, no way mod_skypopen can continue to work. >>> >>> Has been a nice trip, a lot has changed in the Real Time Communication >>> world since first chan_skypiax, then mod_skypopen bridged Skype and open >>> standard calls. >>> >>> mod_skypopen sez: "GOODBYE WORLD!" >>> >>> >>> -giovanni >>> >>> >>> >>> -- >>> >>> Sincerely, >>> >>> Giovanni Maruzzelli >>> OpenTelecom.IT >>> cell: +39 347 266 56 18 >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at 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 at 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 at 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 at 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 > OpenTelecom.IT > cell: +39 347 266 56 18 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Best regards, Volodymyr -------------- next part -------------- An HTML attachment was scrubbed... URL: From gmaruzz at gmail.com Fri Jun 23 13:15:39 2017 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Fri, 23 Jun 2017 15:15:39 +0200 Subject: [Freeswitch-users] SKYPE: mod_skypopen END-OF-LIFE In-Reply-To: References: Message-ID: On 23 June 2017 at 15:07, Volodymyr Fedorov wrote: > Thanks for your work Giovanni, > But I have a question. Did somebody look to ability to bridge webrtc Skype > client and Freeswitch ? > Current version Skype for linux is "just" chromium page is`t it. For chat > protocol exist implementation for Web-skype: > https://github.com/EionRobb/skype4pidgin/tree/master/skypeweb > > if/when there will be an endorsed, legal, free and redistributable way to do it, I will be happy to look into that and try. Definitely not if its legal part is less than clear, and if is not endorsed by Microsoft (eg, not an hack that will stop to work when they change something). That said, good luck to others > Thanks. > > > On Fri, Jun 23, 2017 at 11:29 AM, Giovanni Maruzzelli > wrote: > >> Ciao Mitul !!!! >> >> A friendly and dear hug ! >> -giovanni >> >> >> >> On 23 June 2017 at 11:26, Mitul Limbani wrote: >> >>> Sad to see this die :( >>> >>> Mitul >>> >>> Regards, >>> Mitul Limbani, >>> Business Head, >>> Enterux Solutions Pvt. Ltd. >>> 110 Reena Complex, Opp. Nathani Steel, >>> Vidyavihar (W), Mumbai - 400 086. India >>> http://www.enterux.com/ >>> http://www.entvoice.com/ >>> email: mitul at enterux.in >>> DID: +91-22-71967196 <+91%2022%207196%207196> >>> Cell: +91-9820332422 <+91%2098203%2032422> >>> >>> On Fri, Jun 23, 2017 at 2:16 PM, Giovanni Maruzzelli >>> wrote: >>> >>>> Hello FreeSWITCHers, >>>> >>>> >>>> Microsoft decided to kill all old Skype clients July the 1st, and new >>>> clients do not support APIs to interact with calls. >>>> >>>> So, no way mod_skypopen can continue to work. >>>> >>>> Has been a nice trip, a lot has changed in the Real Time Communication >>>> world since first chan_skypiax, then mod_skypopen bridged Skype and open >>>> standard calls. >>>> >>>> mod_skypopen sez: "GOODBYE WORLD!" >>>> >>>> >>>> -giovanni >>>> >>>> >>>> >>>> -- >>>> >>>> Sincerely, >>>> >>>> Giovanni Maruzzelli >>>> OpenTelecom.IT >>>> cell: +39 347 266 56 18 >>>> >>>> ____________________________________________________________ >>>> _____________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at 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 at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>> switch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at 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 at 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 >> OpenTelecom.IT >> cell: +39 347 266 56 18 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Best regards, > Volodymyr > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at 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 OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: From yu at yu-boot.ru Fri Jun 23 13:40:07 2017 From: yu at yu-boot.ru (Yu Boot) Date: Fri, 23 Jun 2017 16:40:07 +0300 Subject: [Freeswitch-users] Include text file in XML dialplan? In-Reply-To: References: <97617996-90a5-9a30-0497-67e06d941e27@yu-boot.ru> Message-ID: <0949b6f9-ee93-e5e0-9276-a65eb7a6f488@yu-boot.ru> I want something like this: default.xml: .... >>>>> include file.xml here <<<< .... Is it possible? 23.06.2017 15:40, Giovanni Maruzzelli пишет: > all XML dialplan and configs are based on including files. Obviously > the included files must be XML... > > On 23 June 2017 at 13:52, Yu Boot > wrote: > > Is it possible to just include arbitrary text file to some place > in XML dialplan? I want to add or remove some extension and > condition entries automatically, without parsing and building > entire dialplan XML. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at 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 > OpenTelecom.IT > cell: +39 347 266 56 18 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From gmaruzz at gmail.com Fri Jun 23 13:46:51 2017 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Fri, 23 Jun 2017 15:46:51 +0200 Subject: [Freeswitch-users] Include text file in XML dialplan? In-Reply-To: <0949b6f9-ee93-e5e0-9276-a65eb7a6f488@yu-boot.ru> References: <97617996-90a5-9a30-0497-67e06d941e27@yu-boot.ru> <0949b6f9-ee93-e5e0-9276-a65eb7a6f488@yu-boot.ru> Message-ID: On 23 June 2017 at 15:40, Yu Boot wrote: > I want something like this: > > default.xml: > > > > > > .... > > > > > > > >>>>> include file.xml here <<<< > > > > > > > .... > > > > > > > Is it possible? > is already like that. Also, being XML, is inherently structured. What I cannot understand is what you think you gain from this. Eg: it will not be automatically reloaded if you change the included files, you must "reloadxml" . Same would be if you edit the "already there" files... you would need to "reloadxml" > > 23.06.2017 15:40, Giovanni Maruzzelli пишет: > > all XML dialplan and configs are based on including files. Obviously the > included files must be XML... > > On 23 June 2017 at 13:52, Yu Boot wrote: > >> Is it possible to just include arbitrary text file to some place in XML >> dialplan? I want to add or remove some extension and condition entries >> automatically, without parsing and building entire dialplan XML. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at 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 > OpenTelecom.IT > cell: +39 347 266 56 18 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://confluence.freeswitch.orghttp://www.cluecon.com > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at 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 OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: From krice at freeswitch.org Fri Jun 23 14:42:39 2017 From: krice at freeswitch.org (Ken Rice) Date: Fri, 23 Jun 2017 09:42:39 -0500 Subject: [Freeswitch-users] Include text file in XML dialplan? In-Reply-To: <0949b6f9-ee93-e5e0-9276-a65eb7a6f488@yu-boot.ru> References: <97617996-90a5-9a30-0497-67e06d941e27@yu-boot.ru> <0949b6f9-ee93-e5e0-9276-a65eb7a6f488@yu-boot.ru> Message-ID: Why you can include files via the X-PRE include directive, this is not dynamic. You must call reloadxml everytime to get it to reload/parse the dialplan. To do dynamic Dialplan you should look at mod_xml_curl Mr West has a video at https://www.youtube.com/watch?v=IxhLXATa1Ss describing how it works and documentation on confluence at https://freeswitch.org/confluence/display/FREESWITCH/mod_xml_curl From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Yu Boot Sent: Friday, June 23, 2017 8:40 AM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Include text file in XML dialplan? I want something like this: default.xml: .... >>>>> include file.xml here <<<< .... Is it possible? 23.06.2017 15:40, Giovanni Maruzzelli пишет: all XML dialplan and configs are based on including files. Obviously the included files must be XML... On 23 June 2017 at 13:52, Yu Boot > wrote: Is it possible to just include arbitrary text file to some place in XML dialplan? I want to add or remove some extension and condition entries automatically, without parsing and building entire dialplan XML. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at 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 at 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 OpenTelecom.IT cell: +39 347 266 56 18 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From krice at freeswitch.org Fri Jun 23 14:44:21 2017 From: krice at freeswitch.org (Ken Rice) Date: Fri, 23 Jun 2017 09:44:21 -0500 Subject: [Freeswitch-users] Verto specification In-Reply-To: References: Message-ID: There is already work in progress to remove jquery as dep for the base verto lib… if you want to contribute help there check out the branch in git bugfix/FS-10028-refactor-verto-lib-to-remove-jquery From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Luca Pradovera Sent: Friday, June 23, 2017 7:39 AM To: FreeSWITCH Users Help Subject: [Freeswitch-users] Verto specification Hello, one of our clients has a requirement to use Verto in a mobile application (outside of a WebView). That means the JQuery dependency can't be met, and we need to figure out a way around it. I as thinking of just reimplementing the signaling plane without any JQuery dependency and taking it from there. Is the actual JSON protocol documented anywhere? I can also do a Verto Communicator install and sniff traffic, but I was looking if there was anything more structured. Alternatively, does anyone have better ideas to offer? Thanks! Best regards, Luca Pradovera -------------- next part -------------- An HTML attachment was scrubbed... URL: From borik.internet at gmail.com Fri Jun 23 12:09:55 2017 From: borik.internet at gmail.com (Dmitriy Borisov) Date: Fri, 23 Jun 2017 12:09:55 +0000 Subject: [Freeswitch-users] nat-options-ping algorithm Message-ID: Hi, All! I have next topology: PSTN <--- FreeSWITCH <--- channel ---> SIP-users This channel is periodically broken (for 1-10 minutes). Dialplan invoked on incoming ring have 13 endpoints, 10sec to each of 12 first endpoint, and transfer to mobile terminal via PSTN. When channel working good all is fine, but when channel is broken, we have longterm timeout to each endpoint until each of endpoint do not gone to UNREGISTER state. Endpoint gone to UNREGISTER state when REGISTER is expired, but this term is too long for me. I have turn on nat-options-ping option in sip profile, but each endpoint after unsuccessful ping don't gone to UNREGISTER state. When channel is originated to such endpoint it have lifecycle 3sec long and it have ended with NORMAL_TEMPORARY_FAILURE hangup cause. 3 secs by 12 endpoints - too long waiting. When endpoint does not registered - near zero second and ended with USER_NOT_REGISTERED hangup cause. What and where I will must modify to achieve near zero secs term of life channel to inaccessible endpoints? Thanx! -------------- next part -------------- An HTML attachment was scrubbed... URL: From luis.daniel.lucio at gmail.com Fri Jun 23 12:04:43 2017 From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz) Date: Fri, 23 Jun 2017 08:04:43 -0400 Subject: [Freeswitch-users] Include text file in XML dialplan? In-Reply-To: <97617996-90a5-9a30-0497-67e06d941e27@yu-boot.ru> References: <97617996-90a5-9a30-0497-67e06d941e27@yu-boot.ru> Message-ID: Lua or curl is what you need Le 23 juin 2017 7:52 AM, "Yu Boot" a écrit : > Is it possible to just include arbitrary text file to some place in XML > dialplan? I want to add or remove some extension and condition entries > automatically, without parsing and building entire dialplan XML. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From gmaruzz at gmail.com Fri Jun 23 15:54:09 2017 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Fri, 23 Jun 2017 17:54:09 +0200 Subject: [Freeswitch-users] nat-options-ping algorithm In-Reply-To: References: Message-ID: On 23 June 2017 at 14:09, Dmitriy Borisov wrote: > Hi, All! > > I have next topology: PSTN <--- FreeSWITCH <--- channel ---> SIP-users > > This channel is periodically broken (for 1-10 minutes). Dialplan invoked > on incoming ring have 13 endpoints, 10sec to each of 12 first endpoint, and > transfer to mobile terminal via PSTN. When channel working good all is > fine, but when channel is broken, we have longterm timeout to each endpoint > until each of endpoint do not gone to UNREGISTER state. Endpoint gone to > UNREGISTER state when REGISTER is expired, but this term is too long for > me. I have turn on nat-options-ping option in sip profile, but each > endpoint after unsuccessful ping don't gone to UNREGISTER state. When > channel is originated to such endpoint it have lifecycle 3sec long and it > have ended with NORMAL_TEMPORARY_FAILURE hangup cause. 3 secs by 12 > endpoints - too long waiting. When endpoint does not registered - near zero > second and ended with USER_NOT_REGISTERED hangup cause. What and where I > will must modify to achieve near zero secs term of life channel to > inaccessible endpoints? > what is "channel"? Also, the mail is not very clear. No problem your language is not English. Use much more words, examples, and let us understand more. -giovanni > > Thanx! > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at 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 OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: From deepikay at iiitd.ac.in Fri Jun 23 16:02:09 2017 From: deepikay at iiitd.ac.in (Deepika Yadav) Date: Fri, 23 Jun 2017 21:32:09 +0530 Subject: [Freeswitch-users] Freeswitch API access Message-ID: Hi, I have a scenario where multiple conferences are established among users. This is achieved through python programs over ESL. For each conference, a new thread is created. So, should I also use locks before firing originate calls APIs? Regards Deepika https://www.iiitd.edu.in/~deepikay/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From vma at vallimamod.org Fri Jun 23 16:05:24 2017 From: vma at vallimamod.org (Vallimamod Abdullah) Date: Fri, 23 Jun 2017 18:05:24 +0200 Subject: [Freeswitch-users] nat-options-ping algorithm In-Reply-To: References: Message-ID: <7879EBDD-CBF8-4EA3-A449-D2FFCBAD5717@vallimamod.org> Hi, You may find the profile param unregister-on-options-fail useful. Best Regards, -- Vallimamod Abdullah SIP Solutions vma at sipsolutions.fr . > On 23 Jun 2017, at 14:09, Dmitriy Borisov wrote: > > Hi, All! > > I have next topology: PSTN <--- FreeSWITCH <--- channel ---> SIP-users > > This channel is periodically broken (for 1-10 minutes). Dialplan invoked on incoming ring have 13 endpoints, 10sec to each of 12 first endpoint, and transfer to mobile terminal via PSTN. When channel working good all is fine, but when channel is broken, we have longterm timeout to each endpoint until each of endpoint do not gone to UNREGISTER state. Endpoint gone to UNREGISTER state when REGISTER is expired, but this term is too long for me. I have turn on nat-options-ping option in sip profile, but each endpoint after unsuccessful ping don't gone to UNREGISTER state. When channel is originated to such endpoint it have lifecycle 3sec long and it have ended with NORMAL_TEMPORARY_FAILURE hangup cause. 3 secs by 12 endpoints - too long waiting. When endpoint does not registered - near zero second and ended with USER_NOT_REGISTERED hangup cause. What and where I will must modify to achieve near zero secs term of life channel to inaccessible endpoints? > > Thanx! > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From gmaruzz at gmail.com Fri Jun 23 16:07:20 2017 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Fri, 23 Jun 2017 18:07:20 +0200 Subject: [Freeswitch-users] Freeswitch API access In-Reply-To: References: Message-ID: On 23 June 2017 at 18:02, Deepika Yadav wrote: > Hi, > > I have a scenario where multiple conferences are established among users. > This is achieved through python programs over ESL. For each conference, a > new thread is created. So, should I also use locks before firing originate > calls APIs? > A new thread is created where by who? A new Python thread in your app? Which locks? On what? And why? You do not need any interprocess thing on FreeSWITCH side. FS will spawn internal threads to make outbound calls in parallel... -giovanni > > Regards > Deepika > https://www.iiitd.edu.in/~deepikay/ > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at 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 OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: From deepikay at iiitd.ac.in Fri Jun 23 16:18:18 2017 From: deepikay at iiitd.ac.in (Deepika Yadav) Date: Fri, 23 Jun 2017 21:48:18 +0530 Subject: [Freeswitch-users] Freeswitch API access In-Reply-To: References: Message-ID: Yes, there is python based server that receives requests from Android apps. Based on a particular request, it initiates a conference call among a set of users over cellular network. As soon as a new request comes, the server starts a new thread that listens to Freeswitch events required for taking some actions for this conference. I was just worried about the case where multiple requests come making the server call multiple "originate API" with small gap leading to some kind of problem at the Freeswitch side. Regards, Deepika On Fri, Jun 23, 2017 at 9:37 PM, Giovanni Maruzzelli wrote: > > > On 23 June 2017 at 18:02, Deepika Yadav wrote: > >> Hi, >> >> I have a scenario where multiple conferences are established among users. >> This is achieved through python programs over ESL. For each conference, a >> new thread is created. So, should I also use locks before firing originate >> calls APIs? >> > > A new thread is created where by who? A new Python thread in your app? > > Which locks? On what? And why? > > You do not need any interprocess thing on FreeSWITCH side. FS will spawn > internal threads to make outbound calls in parallel... > > -giovanni > > > > >> >> Regards >> Deepika >> https://www.iiitd.edu.in/~deepikay/ >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at 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 > OpenTelecom.IT > cell: +39 347 266 56 18 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Regards Deepika https://www.iiitd.edu.in/~deepikay/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From deepikay at iiitd.ac.in Fri Jun 23 16:19:15 2017 From: deepikay at iiitd.ac.in (Deepika Yadav) Date: Fri, 23 Jun 2017 21:49:15 +0530 Subject: [Freeswitch-users] Freeswitch API access In-Reply-To: References: Message-ID: Here the locks are simple Python locks to safeguard the execution of API. On Fri, Jun 23, 2017 at 9:48 PM, Deepika Yadav wrote: > Yes, there is python based server that receives requests from Android > apps. Based on a particular request, it initiates a conference call among a > set of users over cellular network. As soon as a new request comes, the > server starts a new thread that listens to Freeswitch events required for > taking some actions for this conference. > > I was just worried about the case where multiple requests come making the > server call multiple "originate API" with small gap leading to some kind of > problem at the Freeswitch side. > > Regards, > Deepika > > On Fri, Jun 23, 2017 at 9:37 PM, Giovanni Maruzzelli > wrote: > >> >> >> On 23 June 2017 at 18:02, Deepika Yadav wrote: >> >>> Hi, >>> >>> I have a scenario where multiple conferences are established among >>> users. This is achieved through python programs over ESL. For each >>> conference, a new thread is created. So, should I also use locks before >>> firing originate calls APIs? >>> >> >> A new thread is created where by who? A new Python thread in your app? >> >> Which locks? On what? And why? >> >> You do not need any interprocess thing on FreeSWITCH side. FS will spawn >> internal threads to make outbound calls in parallel... >> >> -giovanni >> >> >> >> >>> >>> Regards >>> Deepika >>> https://www.iiitd.edu.in/~deepikay/ >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at 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 at 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 >> OpenTelecom.IT >> cell: +39 347 266 56 18 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Regards > Deepika > https://www.iiitd.edu.in/~deepikay/ > -- Regards Deepika https://www.iiitd.edu.in/~deepikay/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From gmaruzz at gmail.com Fri Jun 23 16:25:46 2017 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Fri, 23 Jun 2017 18:25:46 +0200 Subject: [Freeswitch-users] Freeswitch API access In-Reply-To: References: Message-ID: On 23 June 2017 at 18:19, Deepika Yadav wrote: > Here the locks are simple Python locks to safeguard the execution of API. > > On Fri, Jun 23, 2017 at 9:48 PM, Deepika Yadav > wrote: > >> Yes, there is python based server that receives requests from Android >> apps. Based on a particular request, it initiates a conference call among a >> set of users over cellular network. As soon as a new request comes, the >> server starts a new thread that listens to Freeswitch events required for >> taking some actions for this conference. >> >> I was just worried about the case where multiple requests come making the >> server call multiple "originate API" with small gap leading to some kind of >> problem at the Freeswitch side. >> > No problems at all. FreeSWITCH is a massively threaded server, each thing is in its own thread, no need to serialize them at all. -giovanni > >> Regards, >> Deepika >> >> On Fri, Jun 23, 2017 at 9:37 PM, Giovanni Maruzzelli >> wrote: >> >>> >>> >>> On 23 June 2017 at 18:02, Deepika Yadav wrote: >>> >>>> Hi, >>>> >>>> I have a scenario where multiple conferences are established among >>>> users. This is achieved through python programs over ESL. For each >>>> conference, a new thread is created. So, should I also use locks before >>>> firing originate calls APIs? >>>> >>> >>> A new thread is created where by who? A new Python thread in your app? >>> >>> Which locks? On what? And why? >>> >>> You do not need any interprocess thing on FreeSWITCH side. FS will spawn >>> internal threads to make outbound calls in parallel... >>> >>> -giovanni >>> >>> >>> >>> >>>> >>>> Regards >>>> Deepika >>>> https://www.iiitd.edu.in/~deepikay/ >>>> >>>> ____________________________________________________________ >>>> _____________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at 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 at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>> switch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> >>> -- >>> >>> Sincerely, >>> >>> Giovanni Maruzzelli >>> OpenTelecom.IT >>> cell: +39 347 266 56 18 >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at 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 at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Regards >> Deepika >> https://www.iiitd.edu.in/~deepikay/ >> > > > > -- > Regards > Deepika > https://www.iiitd.edu.in/~deepikay/ > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at 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 OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: From deepikay at iiitd.ac.in Fri Jun 23 16:30:33 2017 From: deepikay at iiitd.ac.in (Deepika Yadav) Date: Fri, 23 Jun 2017 22:00:33 +0530 Subject: [Freeswitch-users] Freeswitch API access In-Reply-To: References: Message-ID: Thanks for the information. Regards, Deepika On Fri, Jun 23, 2017 at 9:55 PM, Giovanni Maruzzelli wrote: > > > On 23 June 2017 at 18:19, Deepika Yadav wrote: > >> Here the locks are simple Python locks to safeguard the execution of API. >> >> On Fri, Jun 23, 2017 at 9:48 PM, Deepika Yadav >> wrote: >> >>> Yes, there is python based server that receives requests from Android >>> apps. Based on a particular request, it initiates a conference call among a >>> set of users over cellular network. As soon as a new request comes, the >>> server starts a new thread that listens to Freeswitch events required for >>> taking some actions for this conference. >>> >>> I was just worried about the case where multiple requests come making >>> the server call multiple "originate API" with small gap leading to some >>> kind of problem at the Freeswitch side. >>> >> > > No problems at all. FreeSWITCH is a massively threaded server, each thing > is in its own thread, no need to serialize them at all. > > -giovanni > > > > > > > > > >> >>> Regards, >>> Deepika >>> >>> On Fri, Jun 23, 2017 at 9:37 PM, Giovanni Maruzzelli >>> wrote: >>> >>>> >>>> >>>> On 23 June 2017 at 18:02, Deepika Yadav wrote: >>>> >>>>> Hi, >>>>> >>>>> I have a scenario where multiple conferences are established among >>>>> users. This is achieved through python programs over ESL. For each >>>>> conference, a new thread is created. So, should I also use locks before >>>>> firing originate calls APIs? >>>>> >>>> >>>> A new thread is created where by who? A new Python thread in your app? >>>> >>>> Which locks? On what? And why? >>>> >>>> You do not need any interprocess thing on FreeSWITCH side. FS will >>>> spawn internal threads to make outbound calls in parallel... >>>> >>>> -giovanni >>>> >>>> >>>> >>>> >>>>> >>>>> Regards >>>>> Deepika >>>>> https://www.iiitd.edu.in/~deepikay/ >>>>> >>>>> ____________________________________________________________ >>>>> _____________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at 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 at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>>> switch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> >>>> -- >>>> >>>> Sincerely, >>>> >>>> Giovanni Maruzzelli >>>> OpenTelecom.IT >>>> cell: +39 347 266 56 18 >>>> >>>> ____________________________________________________________ >>>> _____________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at 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 at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>> switch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> >>> -- >>> Regards >>> Deepika >>> https://www.iiitd.edu.in/~deepikay/ >>> >> >> >> >> -- >> Regards >> Deepika >> https://www.iiitd.edu.in/~deepikay/ >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at 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 > OpenTelecom.IT > cell: +39 347 266 56 18 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Regards Deepika https://www.iiitd.edu.in/~deepikay/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From gmaruzz at gmail.com Fri Jun 23 16:35:30 2017 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Fri, 23 Jun 2017 18:35:30 +0200 Subject: [Freeswitch-users] Freeswitch API access In-Reply-To: References: Message-ID: On 23 June 2017 at 18:30, Deepika Yadav wrote: > Thanks for the information. > btw, Deepika, I had a look at your web page at IIT (I gave a little conference there many years ago, iirc). Maybe you can find the "Freedomfone" project interesting... http://freedomfone.org/ Is probably a little old but... Best wishes for your PHD, and please let us know how it goes, etc -giovanni > > Regards, > Deepika > > On Fri, Jun 23, 2017 at 9:55 PM, Giovanni Maruzzelli > wrote: > >> >> >> On 23 June 2017 at 18:19, Deepika Yadav wrote: >> >>> Here the locks are simple Python locks to safeguard the execution of API. >>> >>> On Fri, Jun 23, 2017 at 9:48 PM, Deepika Yadav >>> wrote: >>> >>>> Yes, there is python based server that receives requests from Android >>>> apps. Based on a particular request, it initiates a conference call among a >>>> set of users over cellular network. As soon as a new request comes, the >>>> server starts a new thread that listens to Freeswitch events required for >>>> taking some actions for this conference. >>>> >>>> I was just worried about the case where multiple requests come making >>>> the server call multiple "originate API" with small gap leading to some >>>> kind of problem at the Freeswitch side. >>>> >>> >> >> No problems at all. FreeSWITCH is a massively threaded server, each thing >> is in its own thread, no need to serialize them at all. >> >> -giovanni >> >> >> >> >> >> >> >> >> >>> >>>> Regards, >>>> Deepika >>>> >>>> On Fri, Jun 23, 2017 at 9:37 PM, Giovanni Maruzzelli >>> > wrote: >>>> >>>>> >>>>> >>>>> On 23 June 2017 at 18:02, Deepika Yadav wrote: >>>>> >>>>>> Hi, >>>>>> >>>>>> I have a scenario where multiple conferences are established among >>>>>> users. This is achieved through python programs over ESL. For each >>>>>> conference, a new thread is created. So, should I also use locks before >>>>>> firing originate calls APIs? >>>>>> >>>>> >>>>> A new thread is created where by who? A new Python thread in your app? >>>>> >>>>> Which locks? On what? And why? >>>>> >>>>> You do not need any interprocess thing on FreeSWITCH side. FS will >>>>> spawn internal threads to make outbound calls in parallel... >>>>> >>>>> -giovanni >>>>> >>>>> >>>>> >>>>> >>>>>> >>>>>> Regards >>>>>> Deepika >>>>>> https://www.iiitd.edu.in/~deepikay/ >>>>>> >>>>>> ____________________________________________________________ >>>>>> _____________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at 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 at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>>>> switch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>> >>>>> >>>>> >>>>> -- >>>>> >>>>> Sincerely, >>>>> >>>>> Giovanni Maruzzelli >>>>> OpenTelecom.IT >>>>> cell: +39 347 266 56 18 >>>>> >>>>> ____________________________________________________________ >>>>> _____________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at 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 at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>>> switch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> >>>> -- >>>> Regards >>>> Deepika >>>> https://www.iiitd.edu.in/~deepikay/ >>>> >>> >>> >>> >>> -- >>> Regards >>> Deepika >>> https://www.iiitd.edu.in/~deepikay/ >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at 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 at 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 >> OpenTelecom.IT >> cell: +39 347 266 56 18 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Regards > Deepika > https://www.iiitd.edu.in/~deepikay/ > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at 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 OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: From yu at yu-boot.ru Fri Jun 23 16:39:07 2017 From: yu at yu-boot.ru (Yu Boot) Date: Fri, 23 Jun 2017 19:39:07 +0300 Subject: [Freeswitch-users] Include text file in XML dialplan? In-Reply-To: References: <97617996-90a5-9a30-0497-67e06d941e27@yu-boot.ru> <0949b6f9-ee93-e5e0-9276-a65eb7a6f488@yu-boot.ru> Message-ID: <51e20354-dfbb-864d-56c1-c333e89dc376@yu-boot.ru> Let me explain. All inbound calls are routed via one single context, where they are parsed by source IP etc. I want to have an option to insert (and delete) rule that all call will be pass BEFORE any other rules like IP parsing. For example, external script is analyzing CDR and "shuts down" any international calls in case of emergency. I can do this easily by including file with dialplan rules in Yate. I want similiar functionality in FS. 23.06.2017 16:46, Giovanni Maruzzelli пишет: > > On 23 June 2017 at 15:40, Yu Boot > wrote: > > I want something like this: > > default.xml: > > > > > > .... > > > > > > > >>>>> include file.xml here <<<< > > > > > > > .... > > > > > > > Is it possible? > > > is already like that. Also, being XML, is inherently structured. > > What I cannot understand is what you think you gain from this. > > Eg: it will not be automatically reloaded if you change the included > files, you must "reloadxml" . Same would be if you edit the "already > there" files... you would need to "reloadxml" > > > > 23.06.2017 15:40, Giovanni Maruzzelli пишет: >> all XML dialplan and configs are based on including files. >> Obviously the included files must be XML... >> >> On 23 June 2017 at 13:52, Yu Boot > > wrote: >> >> Is it possible to just include arbitrary text file to some >> place in XML dialplan? I want to add or remove some extension >> and condition entries automatically, without parsing and >> building entire dialplan XML. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at 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 >> OpenTelecom.IT >> cell: +39 347 266 56 18 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at 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 at 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 at 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 > OpenTelecom.IT > cell: +39 347 266 56 18 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From gmaruzz at gmail.com Fri Jun 23 16:46:46 2017 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Fri, 23 Jun 2017 18:46:46 +0200 Subject: [Freeswitch-users] Include text file in XML dialplan? In-Reply-To: <51e20354-dfbb-864d-56c1-c333e89dc376@yu-boot.ru> References: <97617996-90a5-9a30-0497-67e06d941e27@yu-boot.ru> <0949b6f9-ee93-e5e0-9276-a65eb7a6f488@yu-boot.ru> <51e20354-dfbb-864d-56c1-c333e89dc376@yu-boot.ru> Message-ID: On 23 June 2017 at 18:39, Yu Boot wrote: > Let me explain. All inbound calls are routed via one single context, where > they are parsed by source IP etc. I want to have an option to insert (and > delete) rule that all call will be pass BEFORE any other rules like IP > parsing. For example, external script is analyzing CDR and "shuts down" any > international calls in case of emergency. I can do this easily by including > file with dialplan rules in Yate. I want similiar functionality in FS. > you insert an extension at dialplan beginning, with a catchall expression, and a continue="true" parameter. All calls will pass through this extension, and then continue. You do your things there, possibly with script. eg: .... Also, you may want to study the freeSWITCH books and the confluence documentation site. -giovanni -------------- next part -------------- An HTML attachment was scrubbed... URL: From deepikay at iiitd.ac.in Fri Jun 23 17:16:08 2017 From: deepikay at iiitd.ac.in (Deepika Yadav) Date: Fri, 23 Jun 2017 22:46:08 +0530 Subject: [Freeswitch-users] Freeswitch API access In-Reply-To: References: Message-ID: There is one more 'I' in the institute name IIIT (https://iiitd.ac.in) :). In my PhD work which belongs to the domain of HCI, broadly, I address the challenge of training Community Health Workers, who work in rural parts of India using mobile technology. I actively use Freeswitch in my work. I have built a platform that combines IVR and android app to provide a synchronous discussion forum for these health workers. Its use can be seen in two of the research papers: http://dl.acm.org/citation.cfm?id=3052624&CFID=777518217&CFTOKEN=57957032 http://dl.acm.org/citation.cfm?id=2858585 I find this Freedomfone software very interesting and will definitely explore it for our use cases. Again, thanks for sharing the information. Regards, Deepika On Fri, Jun 23, 2017 at 10:05 PM, Giovanni Maruzzelli wrote: > > > On 23 June 2017 at 18:30, Deepika Yadav wrote: > >> Thanks for the information. >> > > btw, Deepika, I had a look at your web page at IIT (I gave a little > conference there many years ago, iirc). Maybe you can find the > "Freedomfone" project interesting... http://freedomfone.org/ > > Is probably a little old but... > > Best wishes for your PHD, and please let us know how it goes, etc > > -giovanni > > > >> >> Regards, >> Deepika >> >> On Fri, Jun 23, 2017 at 9:55 PM, Giovanni Maruzzelli >> wrote: >> >>> >>> >>> On 23 June 2017 at 18:19, Deepika Yadav wrote: >>> >>>> Here the locks are simple Python locks to safeguard the execution of >>>> API. >>>> >>>> On Fri, Jun 23, 2017 at 9:48 PM, Deepika Yadav >>>> wrote: >>>> >>>>> Yes, there is python based server that receives requests from Android >>>>> apps. Based on a particular request, it initiates a conference call among a >>>>> set of users over cellular network. As soon as a new request comes, the >>>>> server starts a new thread that listens to Freeswitch events required for >>>>> taking some actions for this conference. >>>>> >>>>> I was just worried about the case where multiple requests come making >>>>> the server call multiple "originate API" with small gap leading to some >>>>> kind of problem at the Freeswitch side. >>>>> >>>> >>> >>> No problems at all. FreeSWITCH is a massively threaded server, each >>> thing is in its own thread, no need to serialize them at all. >>> >>> -giovanni >>> >>> >>> >>> >>> >>> >>> >>> >>> >>>> >>>>> Regards, >>>>> Deepika >>>>> >>>>> On Fri, Jun 23, 2017 at 9:37 PM, Giovanni Maruzzelli < >>>>> gmaruzz at gmail.com> wrote: >>>>> >>>>>> >>>>>> >>>>>> On 23 June 2017 at 18:02, Deepika Yadav wrote: >>>>>> >>>>>>> Hi, >>>>>>> >>>>>>> I have a scenario where multiple conferences are established among >>>>>>> users. This is achieved through python programs over ESL. For each >>>>>>> conference, a new thread is created. So, should I also use locks before >>>>>>> firing originate calls APIs? >>>>>>> >>>>>> >>>>>> A new thread is created where by who? A new Python thread in your app? >>>>>> >>>>>> Which locks? On what? And why? >>>>>> >>>>>> You do not need any interprocess thing on FreeSWITCH side. FS will >>>>>> spawn internal threads to make outbound calls in parallel... >>>>>> >>>>>> -giovanni >>>>>> >>>>>> >>>>>> >>>>>> >>>>>>> >>>>>>> Regards >>>>>>> Deepika >>>>>>> https://www.iiitd.edu.in/~deepikay/ >>>>>>> >>>>>>> ____________________________________________________________ >>>>>>> _____________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at 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 at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>>>>> switch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> >>>>>> Sincerely, >>>>>> >>>>>> Giovanni Maruzzelli >>>>>> OpenTelecom.IT >>>>>> cell: +39 347 266 56 18 >>>>>> >>>>>> ____________________________________________________________ >>>>>> _____________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at 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 at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>>>> switch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>> >>>>> >>>>> >>>>> -- >>>>> Regards >>>>> Deepika >>>>> https://www.iiitd.edu.in/~deepikay/ >>>>> >>>> >>>> >>>> >>>> -- >>>> Regards >>>> Deepika >>>> https://www.iiitd.edu.in/~deepikay/ >>>> >>>> ____________________________________________________________ >>>> _____________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at 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 at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>> switch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> >>> -- >>> >>> Sincerely, >>> >>> Giovanni Maruzzelli >>> OpenTelecom.IT >>> cell: +39 347 266 56 18 >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at 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 at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Regards >> Deepika >> https://www.iiitd.edu.in/~deepikay/ >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at 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 > OpenTelecom.IT > cell: +39 347 266 56 18 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Regards Deepika https://www.iiitd.edu.in/~deepikay/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From gmaruzz at gmail.com Fri Jun 23 18:59:35 2017 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Fri, 23 Jun 2017 20:59:35 +0200 Subject: [Freeswitch-users] Freeswitch API access In-Reply-To: References: Message-ID: On 23 June 2017 at 19:16, Deepika Yadav wrote: > There is one more 'I' in the institute name IIIT (https://iiitd.ac.in) > :). In my PhD work which belongs to the domain of HCI, broadly, I address > the challenge of training Community Health Workers, who work in rural parts > of India using mobile technology. I actively use Freeswitch in my work. I > have built a platform that combines IVR and android app to provide a > synchronous discussion forum for these health workers. Its use can be seen > in two of the research papers: > > http://dl.acm.org/citation.cfm?id=3052624&CFID=777518217&CFTOKEN=57957032 > > http://dl.acm.org/citation.cfm?id=2858585 > > I find this Freedomfone software very interesting and will definitely > explore it for our use cases. > > Again, thanks for sharing the information. > Thanks to you Deepika, we're proud and glad FreeSWITCH is used in such a project, and we would like so much to have you presenting it in one of our weekly videoconferences, that are also on youtube. Please, be in contact and write to kathleen at freeswitch.org for our weekly call. -giovanni > > Regards, > Deepika > > On Fri, Jun 23, 2017 at 10:05 PM, Giovanni Maruzzelli > wrote: > >> >> >> On 23 June 2017 at 18:30, Deepika Yadav wrote: >> >>> Thanks for the information. >>> >> >> btw, Deepika, I had a look at your web page at IIT (I gave a little >> conference there many years ago, iirc). Maybe you can find the >> "Freedomfone" project interesting... http://freedomfone.org/ >> >> Is probably a little old but... >> >> Best wishes for your PHD, and please let us know how it goes, etc >> >> -giovanni >> >> >> >>> >>> Regards, >>> Deepika >>> >>> On Fri, Jun 23, 2017 at 9:55 PM, Giovanni Maruzzelli >>> wrote: >>> >>>> >>>> >>>> On 23 June 2017 at 18:19, Deepika Yadav wrote: >>>> >>>>> Here the locks are simple Python locks to safeguard the execution of >>>>> API. >>>>> >>>>> On Fri, Jun 23, 2017 at 9:48 PM, Deepika Yadav >>>>> wrote: >>>>> >>>>>> Yes, there is python based server that receives requests from Android >>>>>> apps. Based on a particular request, it initiates a conference call among a >>>>>> set of users over cellular network. As soon as a new request comes, the >>>>>> server starts a new thread that listens to Freeswitch events required for >>>>>> taking some actions for this conference. >>>>>> >>>>>> I was just worried about the case where multiple requests come making >>>>>> the server call multiple "originate API" with small gap leading to some >>>>>> kind of problem at the Freeswitch side. >>>>>> >>>>> >>>> >>>> No problems at all. FreeSWITCH is a massively threaded server, each >>>> thing is in its own thread, no need to serialize them at all. >>>> >>>> -giovanni >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>>> >>>>>> Regards, >>>>>> Deepika >>>>>> >>>>>> On Fri, Jun 23, 2017 at 9:37 PM, Giovanni Maruzzelli < >>>>>> gmaruzz at gmail.com> wrote: >>>>>> >>>>>>> >>>>>>> >>>>>>> On 23 June 2017 at 18:02, Deepika Yadav >>>>>>> wrote: >>>>>>> >>>>>>>> Hi, >>>>>>>> >>>>>>>> I have a scenario where multiple conferences are established among >>>>>>>> users. This is achieved through python programs over ESL. For each >>>>>>>> conference, a new thread is created. So, should I also use locks before >>>>>>>> firing originate calls APIs? >>>>>>>> >>>>>>> >>>>>>> A new thread is created where by who? A new Python thread in your >>>>>>> app? >>>>>>> >>>>>>> Which locks? On what? And why? >>>>>>> >>>>>>> You do not need any interprocess thing on FreeSWITCH side. FS will >>>>>>> spawn internal threads to make outbound calls in parallel... >>>>>>> >>>>>>> -giovanni >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>>> >>>>>>>> Regards >>>>>>>> Deepika >>>>>>>> https://www.iiitd.edu.in/~deepikay/ >>>>>>>> >>>>>>>> ____________________________________________________________ >>>>>>>> _____________ >>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>> consulting at 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 at lists.freeswitch.org >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>>>>>> switch-users >>>>>>>> http://www.freeswitch.org >>>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> -- >>>>>>> >>>>>>> Sincerely, >>>>>>> >>>>>>> Giovanni Maruzzelli >>>>>>> OpenTelecom.IT >>>>>>> cell: +39 347 266 56 18 >>>>>>> >>>>>>> ____________________________________________________________ >>>>>>> _____________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at 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 at lists.freeswitch.org >>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>>>>> switch-users >>>>>>> http://www.freeswitch.org >>>>>>> >>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> Regards >>>>>> Deepika >>>>>> https://www.iiitd.edu.in/~deepikay/ >>>>>> >>>>> >>>>> >>>>> >>>>> -- >>>>> Regards >>>>> Deepika >>>>> https://www.iiitd.edu.in/~deepikay/ >>>>> >>>>> ____________________________________________________________ >>>>> _____________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at 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 at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>>> switch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> >>>> -- >>>> >>>> Sincerely, >>>> >>>> Giovanni Maruzzelli >>>> OpenTelecom.IT >>>> cell: +39 347 266 56 18 >>>> >>>> ____________________________________________________________ >>>> _____________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at 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 at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>> switch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> >>> -- >>> Regards >>> Deepika >>> https://www.iiitd.edu.in/~deepikay/ >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at 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 at 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 >> OpenTelecom.IT >> cell: +39 347 266 56 18 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Regards > Deepika > https://www.iiitd.edu.in/~deepikay/ > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at 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 OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: From borik.internet at gmail.com Fri Jun 23 19:45:51 2017 From: borik.internet at gmail.com (Dmitriy Borisov) Date: Fri, 23 Jun 2017 19:45:51 +0000 Subject: [Freeswitch-users] nat-options-ping algorithm In-Reply-To: <7879EBDD-CBF8-4EA3-A449-D2FFCBAD5717@vallimamod.org> References: <7879EBDD-CBF8-4EA3-A449-D2FFCBAD5717@vallimamod.org> Message-ID: Abdullah, thank you! I'll go to try, it must be useful пт, 23 июня 2017 г., 19:06 Vallimamod Abdullah : > Hi, > > You may find the profile param unregister-on-options-fail useful. > > Best Regards, > -- > Vallimamod Abdullah > SIP Solutions > vma at sipsolutions.fr > . > > > > On 23 Jun 2017, at 14:09, Dmitriy Borisov > wrote: > > > > Hi, All! > > > > I have next topology: PSTN <--- FreeSWITCH <--- channel ---> SIP-users > > > > This channel is periodically broken (for 1-10 minutes). Dialplan invoked > on incoming ring have 13 endpoints, 10sec to each of 12 first endpoint, and > transfer to mobile terminal via PSTN. When channel working good all is > fine, but when channel is broken, we have longterm timeout to each endpoint > until each of endpoint do not gone to UNREGISTER state. Endpoint gone to > UNREGISTER state when REGISTER is expired, but this term is too long for > me. I have turn on nat-options-ping option in sip profile, but each > endpoint after unsuccessful ping don't gone to UNREGISTER state. When > channel is originated to such endpoint it have lifecycle 3sec long and it > have ended with NORMAL_TEMPORARY_FAILURE hangup cause. 3 secs by 12 > endpoints - too long waiting. When endpoint does not registered - near zero > second and ended with USER_NOT_REGISTERED hangup cause. What and where I > will must modify to achieve near zero secs term of life channel to > inaccessible endpoints? > > > > Thanx! > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From anthony.minessale at gmail.com Fri Jun 23 20:00:36 2017 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 23 Jun 2017 15:00:36 -0500 Subject: [Freeswitch-users] Javascript FileIO In-Reply-To: <2e9e2174.97ca.15cd41840cd.Coremail.xxxman2008@126.com> References: <2e9e2174.97ca.15cd41840cd.Coremail.xxxman2008@126.com> Message-ID: That wiki documentation is from the old spidermonkey js module not the new v8 one. I believe that putting a very large number will not cause any problems as the data read cannot exceed the total bytes in the file so it should not be a problem. You can always do an append loop. Its usually a good practice to have a precise max size to read because what if the file was accidentally 2gb. On Fri, Jun 23, 2017 at 3:36 AM, Raymond wrote: > Hi , > > This will help you . https://wiki.freeswitch.org/wiki/File > > Raymond > > At 2017-06-22 21:49:33, "Faruq Ahmad" wrote: > > Hi, > > I have a FS dialplan that fetches callflow from a JSON file. File is read > using > > var JsonFd = new FileIO(JsonPath, 'r'); > JsonFd.read(2048); > result = JsonFd.data(); > var menu = JSON.parse(result), > > Is there anyway I can read the whole file in one attempt, i.e. when a > smaller size was given to the .read() function it wouldn't read the > complete file. I have increased the value of size for read function however > my concern is in the long run file size might increase and parse would get > an incomplete JSON. > > Is there anyway I can get filesize from FileIO object or detect EOF from > the read buffer to make sure the whole file is loaded? > Also if I increase the size for read fucntion buffer way over the > estimated filesizes, is there any guarantee that no garbage values will be > read from the disk after the EOF? > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at 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 ♬ @anthmfs ♬ @FreeSWITCH ♬ ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ * ClueCon Weekly Development Call ☎ sip:888 at conference.freeswitch.org ☎ +19193869900 https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA -------------- next part -------------- An HTML attachment was scrubbed... URL: From afarooqa at gmail.com Fri Jun 23 22:48:11 2017 From: afarooqa at gmail.com (Faruq Ahmad) Date: Sat, 24 Jun 2017 03:48:11 +0500 Subject: [Freeswitch-users] Javascript FileIO In-Reply-To: References: <2e9e2174.97ca.15cd41840cd.Coremail.xxxman2008@126.com> Message-ID: I consulted this wiki and File is not implemented for v8, only FileIo is available. I ended up doing something similar, var read_size = 2048; do { read_size *=2; var jsFd = new FileIO('/opt/prj_jsons/cfId.json', 'r'); jsFd.read(read_size); jsData = jsFd.data(); jsFd.close(); }while(jsData.length == read_size); its a good idea to put a cap on the read_size, i'll put a cap on after a few Mbs. I tried appending the new data to the incomplete read attempt, but after the first iteration the buffer wont clear and on last iteration data from previous read attempt was concatenated to the .data() output of the last read attempt after the data in file finished. if the data read cannot exceed the total bytes in the file I can get rid of this and safely just put the desired cap size in the read_size. Thanks. On Sat, Jun 24, 2017 at 1:00 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > That wiki documentation is from the old spidermonkey js module not the new > v8 one. > > I believe that putting a very large number will not cause any problems as > the data read cannot exceed the total bytes in the file so it should not be > a problem. > You can always do an append loop. Its usually a good practice to have a > precise max size to read because what if the file was accidentally 2gb. > > On Fri, Jun 23, 2017 at 3:36 AM, Raymond wrote: > >> Hi , >> >> This will help you . https://wiki.freeswitch.org/wiki/File >> >> Raymond >> >> At 2017-06-22 21:49:33, "Faruq Ahmad" wrote: >> >> Hi, >> >> I have a FS dialplan that fetches callflow from a JSON file. File is read >> using >> >> var JsonFd = new FileIO(JsonPath, 'r'); >> JsonFd.read(2048); >> result = JsonFd.data(); >> var menu = JSON.parse(result), >> >> Is there anyway I can read the whole file in one attempt, i.e. when a >> smaller size was given to the .read() function it wouldn't read the >> complete file. I have increased the value of size for read function however >> my concern is in the long run file size might increase and parse would get >> an incomplete JSON. >> >> Is there anyway I can get filesize from FileIO object or detect EOF from >> the read buffer to make sure the whole file is loaded? >> Also if I increase the size for read fucntion buffer way over the >> estimated filesizes, is there any guarantee that no garbage values will be >> read from the disk after the EOF? >> >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at 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 ♬ @anthmfs ♬ @FreeSWITCH ♬ > > ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ > http://twitter.com/FreeSWITCH > ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ > * > > ClueCon Weekly Development Call > ☎ sip:888 at conference.freeswitch.org ☎ +19193869900 <(919)%20386-9900> > > https://www.youtube.com/watch?v=9XXgW34t40s > https://www.youtube.com/watch?v=NLaDpGQuZDA > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From grcamauer at gmail.com Fri Jun 23 22:57:17 2017 From: grcamauer at gmail.com (Guillermo Ruiz Camauer) Date: Fri, 23 Jun 2017 19:57:17 -0300 Subject: [Freeswitch-users] Problem with fs_encode Message-ID: I have created a small utility program based on fs_encode which will take ALL files in a directory and convert them from .WAV to .G729 and .PCMU formats. The utility is called wavBatchEncode and takes just a directory path as an argument. This program was working very well until I upgraded FreeSwitch. Now the program seems to hang when it tries to write out the first converted files. I have to KILL the program from another terminal. It leaves 2 0 byte files with .PCMU and .G729 extension. I have run through a GIT bisect to find where things broke. The last working version is Version 1.6.16 git ae1cdce 2017-04-11. The first broken version is 38621e47bad3b63f03a0a27f6ca9ed92f6969032. I then used GIT DIFF to see if I could see what had happened. There are very few modifications between these two commits: root at fs3:/usr/src/freeswitch.git# git diff ae1cdce 38621e47bad3b63f03a0a27f6ca9ed92f6969032 diff --git a/src/include/switch_module_interfaces.h b/src/include/switch_module_interfaces.h index e0a5c20..7ca027d 100644 --- a/src/include/switch_module_interfaces.h +++ b/src/include/switch_module_interfaces.h @@ -329,6 +329,11 @@ typedef struct switch_mm_s { switch_video_profile_t vprofile; switch_video_encode_speed_t vencspd; uint8_t try_hardware_encoder; + int scale_w; + int scale_h; + switch_img_fmt_t fmt; + char *auth_username; + char *auth_password; } switch_mm_t; /*! an abstract representation of a file handle (some parameters based on compat with libsndfile) */ diff --git a/src/mod/applications/mod_av/avformat.c b/src/mod/applications/mod_av/avformat.c index b944625..4b92801 100644 --- a/src/mod/applications/mod_av/avformat.c +++ b/src/mod/applications/mod_av/avformat.c @@ -906,7 +906,7 @@ SWITCH_STANDARD_APP(record_av_function) char codec_str[256]; const AVCodecDescriptor *desc; - if (!strncmp(data, "rtmp://", 7)) { + if (!strncmp(data, "rtmp://", 7) || !strncmp(data, "rtsp://", 7)) { fmt->video_codec = AV_CODEC_ID_H264; fmt->audio_codec = AV_CODEC_ID_AAC; } @@ -1694,9 +1694,20 @@ static switch_status_t av_file_open(switch_file_handle_t *handle, const char *pa return SWITCH_STATUS_GENERR; } else if (handle->stream_name && (!strcasecmp(handle->stream_name, "rtmp") || !strcasecmp(handle->stream_name, "youtube"))) { format = "flv"; - switch_snprintf(file, sizeof(file), "rtmp://%s", path); + + // meh really silly format for the user / pass libav..... + if (handle->mm.auth_username && handle->mm.auth_password) { + switch_snprintf(file, sizeof(file), "rtmp://%s pubUser=%s pubPasswd=%s flashver=FMLE/3.0", path, handle->mm.auth_username, handle->mm.auth_password); + } else { + switch_snprintf(file, sizeof(file), "rtmp://%s", path); + } + + } else if (handle->stream_name && !strcasecmp(handle->stream_name, "rtsp")) { + format = "rtsp"; + switch_snprintf(file, sizeof(file), "rtsp://%s", path); } + ext++; if ((context = (av_file_context_t *)switch_core_alloc(handle->memory_pool, sizeof(av_file_context_t))) == 0) { @@ -1783,7 +1794,7 @@ static switch_status_t av_file_open(switch_file_handle_t *handle, const char *pa if (fmt->video_codec != AV_CODEC_ID_NONE) { const AVCodecDescriptor *desc; - if ((handle->stream_name && (!strcasecmp(handle->stream_name, "rtmp") || !strcasecmp(handle->stream_name, "youtube")))) { + if ((handle->stream_name && (!strcasecmp(handle->stream_name, "rtmp") || !strcasecmp(handle->stream_name, "rtsp") || !strcasecmp(handle->stream_name, "youtube")))) { if (fmt->video_codec != AV_CODEC_ID_H264 ) { fmt->video_codec = AV_CODEC_ID_H264; // force H264 @@ -2525,6 +2536,7 @@ SWITCH_MODULE_LOAD_FUNCTION(mod_avformat_load) supported_formats[i++] = "av"; supported_formats[i++] = "rtmp"; + supported_formats[i++] = "rtsp"; supported_formats[i++] = "mp4"; supported_formats[i++] = "m4a"; supported_formats[i++] = "mov"; diff --git a/src/mod/applications/mod_av/mod_av.c b/src/mod/applications/mod_av/mod_av.c index 3d3bd82..141fcdc 100644 --- a/src/mod/applications/mod_av/mod_av.c +++ b/src/mod/applications/mod_av/mod_av.c @@ -93,7 +93,7 @@ int mod_av_lockmgr_cb(void **m, enum AVLockOp op) static void log_callback(void *ptr, int level, const char *fmt, va_list vl) { switch_log_level_t switch_level = SWITCH_LOG_DEBUG; - + return; /* naggy messages */ if (level == AV_LOG_DEBUG || level == AV_LOG_WARNING) return; diff --git a/src/switch_core_file.c b/src/switch_core_file.c index 46ee539..aff9442 100644 --- a/src/switch_core_file.c +++ b/src/switch_core_file.c @@ -177,6 +177,14 @@ SWITCH_DECLARE(switch_status_t) switch_core_perform_file_open(const char *file, fh->mm.try_hardware_encoder = switch_true(val); } + if ((val = switch_event_get_header(fh->params, "auth_username"))) { + fh->mm.auth_username = switch_core_strdup(fh->memory_pool, val); + } + + if ((val = switch_event_get_header(fh->params, "auth_password"))) { + fh->mm.auth_password = switch_core_strdup(fh->memory_pool, val); + } + if ((val = switch_event_get_header(fh->params, "fps"))) { float ftmp = atof(val); if (ftmp > 0.0f) { (END) I believe that it is the changes in switch_core_file.c that break my utility program, but I can't be sure. Just o add some detail, I have a Digium TCE400 card in my system which handles the G729 conversion. I have no idea of what those auth_username and auth_password parameters are, but that seems to break things for me. Any idea of what I can do to get things working again? I have created a PasteBin with the code of my fs_encode derived utility: https://pastebin.freeswitch.org/view/a3d64c69 It is pretty short. -- Guillermo Ruiz Camauer -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Fri Jun 23 23:01:09 2017 From: mike at jerris.com (Michael Jerris) Date: Fri, 23 Jun 2017 19:01:09 -0400 Subject: [Freeswitch-users] Problem with fs_encode In-Reply-To: References: Message-ID: can you drop a core file and see where its stuck? > On Jun 23, 2017, at 6:57 PM, Guillermo Ruiz Camauer wrote: > > I have created a small utility program based on fs_encode which will take ALL files in a directory and convert them from .WAV to .G729 and .PCMU formats. The utility is called wavBatchEncode and takes just a directory path as an argument. > > This program was working very well until I upgraded FreeSwitch. Now the program seems to hang when it tries to write out the first converted files. I have to KILL the program from another terminal. It leaves 2 0 byte files with .PCMU and .G729 extension. > > I have run through a GIT bisect to find where things broke. The last working version is Version 1.6.16 git ae1cdce 2017-04-11. > The first broken version is 38621e47bad3b63f03a0a27f6ca9ed92f6969032. > > I then used GIT DIFF to see if I could see what had happened. There are very few modifications between these two commits: > > root at fs3:/usr/src/freeswitch.git# git diff ae1cdce 38621e47bad3b63f03a0a27f6ca9ed92f6969032 > diff --git a/src/include/switch_module_interfaces.h b/src/include/switch_module_interfaces.h > index e0a5c20..7ca027d 100644 > --- a/src/include/switch_module_interfaces.h > +++ b/src/include/switch_module_interfaces.h > @@ -329,6 +329,11 @@ typedef struct switch_mm_s { > switch_video_profile_t vprofile; > switch_video_encode_speed_t vencspd; > uint8_t try_hardware_encoder; > + int scale_w; > + int scale_h; > + switch_img_fmt_t fmt; > + char *auth_username; > + char *auth_password; > } switch_mm_t; > > /*! an abstract representation of a file handle (some parameters based on compat with libsndfile) */ > diff --git a/src/mod/applications/mod_av/avformat.c b/src/mod/applications/mod_av/avformat.c > index b944625..4b92801 100644 > --- a/src/mod/applications/mod_av/avformat.c > +++ b/src/mod/applications/mod_av/avformat.c > @@ -906,7 +906,7 @@ SWITCH_STANDARD_APP(record_av_function) > char codec_str[256]; > const AVCodecDescriptor *desc; > > - if (!strncmp(data, "rtmp://", 7)) { > + if (!strncmp(data, "rtmp://", 7) || !strncmp(data, "rtsp://", 7)) { > fmt->video_codec = AV_CODEC_ID_H264; > fmt->audio_codec = AV_CODEC_ID_AAC; > } > @@ -1694,9 +1694,20 @@ static switch_status_t av_file_open(switch_file_handle_t *handle, const char *pa > return SWITCH_STATUS_GENERR; > } else if (handle->stream_name && (!strcasecmp(handle->stream_name, "rtmp") || !strcasecmp(handle->stream_name, "youtube"))) { > format = "flv"; > - switch_snprintf(file, sizeof(file), "rtmp://%s", path); > + > + // meh really silly format for the user / pass libav..... > + if (handle->mm.auth_username && handle->mm.auth_password) { > + switch_snprintf(file, sizeof(file), "rtmp://%s pubUser=%s pubPasswd=%s flashver=FMLE/3.0", path, handle->mm.auth_username, handle->mm.auth_password); > + } else { > + switch_snprintf(file, sizeof(file), "rtmp://%s", path); > + } > + > + } else if (handle->stream_name && !strcasecmp(handle->stream_name, "rtsp")) { > + format = "rtsp"; > + switch_snprintf(file, sizeof(file), "rtsp://%s", path); > } > > + > ext++; > > if ((context = (av_file_context_t *)switch_core_alloc(handle->memory_pool, sizeof(av_file_context_t))) == 0) { > @@ -1783,7 +1794,7 @@ static switch_status_t av_file_open(switch_file_handle_t *handle, const char *pa > if (fmt->video_codec != AV_CODEC_ID_NONE) { > const AVCodecDescriptor *desc; > > - if ((handle->stream_name && (!strcasecmp(handle->stream_name, "rtmp") || !strcasecmp(handle->stream_name, "youtube")))) { > + if ((handle->stream_name && (!strcasecmp(handle->stream_name, "rtmp") || !strcasecmp(handle->stream_name, "rtsp") || !strcasecmp(handle->stream_name, "youtube")))) { > > if (fmt->video_codec != AV_CODEC_ID_H264 ) { > fmt->video_codec = AV_CODEC_ID_H264; // force H264 > @@ -2525,6 +2536,7 @@ SWITCH_MODULE_LOAD_FUNCTION(mod_avformat_load) > > supported_formats[i++] = "av"; > supported_formats[i++] = "rtmp"; > + supported_formats[i++] = "rtsp"; > supported_formats[i++] = "mp4"; > supported_formats[i++] = "m4a"; > supported_formats[i++] = "mov"; > diff --git a/src/mod/applications/mod_av/mod_av.c b/src/mod/applications/mod_av/mod_av.c > index 3d3bd82..141fcdc 100644 > --- a/src/mod/applications/mod_av/mod_av.c > +++ b/src/mod/applications/mod_av/mod_av.c > @@ -93,7 +93,7 @@ int mod_av_lockmgr_cb(void **m, enum AVLockOp op) > static void log_callback(void *ptr, int level, const char *fmt, va_list vl) > { > switch_log_level_t switch_level = SWITCH_LOG_DEBUG; > - > + return; > /* naggy messages */ > if (level == AV_LOG_DEBUG || level == AV_LOG_WARNING) return; > > diff --git a/src/switch_core_file.c b/src/switch_core_file.c > index 46ee539..aff9442 100644 > --- a/src/switch_core_file.c > +++ b/src/switch_core_file.c > @@ -177,6 +177,14 @@ SWITCH_DECLARE(switch_status_t) switch_core_perform_file_open(const char *file, > fh->mm.try_hardware_encoder = switch_true(val); > } > > + if ((val = switch_event_get_header(fh->params, "auth_username"))) { > + fh->mm.auth_username = switch_core_strdup(fh->memory_pool, val); > + } > + > + if ((val = switch_event_get_header(fh->params, "auth_password"))) { > + fh->mm.auth_password = switch_core_strdup(fh->memory_pool, val); > + } > + > if ((val = switch_event_get_header(fh->params, "fps"))) { > float ftmp = atof(val); > if (ftmp > 0.0f) { > (END) > > > I believe that it is the changes in switch_core_file.c that break my utility program, but I can't be sure. Just o add some detail, I have a Digium TCE400 card in my system which handles the G729 conversion. > > I have no idea of what those auth_username and auth_password parameters are, but that seems to break things for me. Any idea of what I can do to get things working again? > > I have created a PasteBin with the code of my fs_encode derived utility: https://pastebin.freeswitch.org/view/a3d64c69 It is pretty short. > -------------- next part -------------- An HTML attachment was scrubbed... URL: From anthony.minessale at gmail.com Fri Jun 23 23:03:27 2017 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 23 Jun 2017 18:03:27 -0500 Subject: [Freeswitch-users] Problem with fs_encode In-Reply-To: References: Message-ID: Also does removing just that change in switch_core_file.c from latest version actually make it work? It seems unlikely the next line in the file looking for FPS field is no different really. On Fri, Jun 23, 2017 at 6:01 PM, Michael Jerris wrote: > can you drop a core file and see where its stuck? > > On Jun 23, 2017, at 6:57 PM, Guillermo Ruiz Camauer > wrote: > > I have created a small utility program based on fs_encode which will take > ALL files in a directory and convert them from .WAV to .G729 and .PCMU > formats. The utility is called wavBatchEncode and takes just a directory > path as an argument. > > This program was working very well until I upgraded FreeSwitch. Now the > program seems to hang when it tries to write out the first converted > files. I have to KILL the program from another terminal. It leaves 2 0 > byte files with .PCMU and .G729 extension. > > I have run through a GIT bisect to find where things broke. The last > working version is Version 1.6.16 git ae1cdce 2017-04-11. > The first broken version is 38621e47bad3b63f03a0a27f6ca9ed92f6969032. > > I then used GIT DIFF to see if I could see what had happened. There are > very few modifications between these two commits: > > root at fs3:/usr/src/freeswitch.git# git diff ae1cdce > 38621e47bad3b63f03a0a27f6ca9ed92f6969032 > diff --git a/src/include/switch_module_interfaces.h > b/src/include/switch_module_interfaces.h > index e0a5c20..7ca027d 100644 > --- a/src/include/switch_module_interfaces.h > +++ b/src/include/switch_module_interfaces.h > @@ -329,6 +329,11 @@ typedef struct switch_mm_s { > switch_video_profile_t vprofile; > switch_video_encode_speed_t vencspd; > uint8_t try_hardware_encoder; > + int scale_w; > + int scale_h; > + switch_img_fmt_t fmt; > + char *auth_username; > + char *auth_password; > } switch_mm_t; > > /*! an abstract representation of a file handle (some parameters based on > compat with libsndfile) */ > diff --git a/src/mod/applications/mod_av/avformat.c > b/src/mod/applications/mod_av/avformat.c > index b944625..4b92801 100644 > --- a/src/mod/applications/mod_av/avformat.c > +++ b/src/mod/applications/mod_av/avformat.c > @@ -906,7 +906,7 @@ SWITCH_STANDARD_APP(record_av_function) > char codec_str[256]; > const AVCodecDescriptor *desc; > > - if (!strncmp(data, "rtmp://", 7)) { > + if (!strncmp(data, "rtmp://", 7) || !strncmp(data, > "rtsp://", 7)) { > fmt->video_codec = AV_CODEC_ID_H264; > fmt->audio_codec = AV_CODEC_ID_AAC; > } > @@ -1694,9 +1694,20 @@ static switch_status_t av_file_open(switch_file_handle_t > *handle, const char *pa > return SWITCH_STATUS_GENERR; > } else if (handle->stream_name && (!strcasecmp(handle->stream_name, > "rtmp") || !strcasecmp(handle->stream_name, "youtube"))) { > format = "flv"; > - switch_snprintf(file, sizeof(file), "rtmp://%s", path); > + > + // meh really silly format for the user / pass libav..... > + if (handle->mm.auth_username && handle->mm.auth_password) > { > + switch_snprintf(file, sizeof(file), "rtmp://%s > pubUser=%s pubPasswd=%s flashver=FMLE/3.0", path, handle->mm.auth_username, > handle->mm.auth_password); > + } else { > + switch_snprintf(file, sizeof(file), "rtmp://%s", > path); > + } > + > + } else if (handle->stream_name && !strcasecmp(handle->stream_name, > "rtsp")) { > + format = "rtsp"; > + switch_snprintf(file, sizeof(file), "rtsp://%s", path); > } > > + > ext++; > > if ((context = (av_file_context_t *)switch_core_alloc(handle->memory_pool, > sizeof(av_file_context_t))) == 0) { > @@ -1783,7 +1794,7 @@ static switch_status_t av_file_open(switch_file_handle_t > *handle, const char *pa > if (fmt->video_codec != AV_CODEC_ID_NONE) { > const AVCodecDescriptor *desc; > > - if ((handle->stream_name && (!strcasecmp(handle->stream_name, > "rtmp") || !strcasecmp(handle->stream_name, "youtube")))) { > + if ((handle->stream_name && (!strcasecmp(handle->stream_name, > "rtmp") || !strcasecmp(handle->stream_name, "rtsp") || > !strcasecmp(handle->stream_name, "youtube")))) { > > if (fmt->video_codec != AV_CODEC_ID_H264 ) { > fmt->video_codec = AV_CODEC_ID_H264; // > force H264 > @@ -2525,6 +2536,7 @@ SWITCH_MODULE_LOAD_FUNCTION(mod_avformat_load) > > supported_formats[i++] = "av"; > supported_formats[i++] = "rtmp"; > + supported_formats[i++] = "rtsp"; > supported_formats[i++] = "mp4"; > supported_formats[i++] = "m4a"; > supported_formats[i++] = "mov"; > diff --git a/src/mod/applications/mod_av/mod_av.c > b/src/mod/applications/mod_av/mod_av.c > index 3d3bd82..141fcdc 100644 > --- a/src/mod/applications/mod_av/mod_av.c > +++ b/src/mod/applications/mod_av/mod_av.c > @@ -93,7 +93,7 @@ int mod_av_lockmgr_cb(void **m, enum AVLockOp op) > static void log_callback(void *ptr, int level, const char *fmt, va_list > vl) > { > switch_log_level_t switch_level = SWITCH_LOG_DEBUG; > - > + return; > /* naggy messages */ > if (level == AV_LOG_DEBUG || level == AV_LOG_WARNING) return; > > diff --git a/src/switch_core_file.c b/src/switch_core_file.c > index 46ee539..aff9442 100644 > --- a/src/switch_core_file.c > +++ b/src/switch_core_file.c > @@ -177,6 +177,14 @@ SWITCH_DECLARE(switch_status_t) > switch_core_perform_file_open(const char *file, > fh->mm.try_hardware_encoder = switch_true(val); > } > > + if ((val = switch_event_get_header(fh->params, > "auth_username"))) { > + fh->mm.auth_username = > switch_core_strdup(fh->memory_pool, val); > + } > + > + if ((val = switch_event_get_header(fh->params, > "auth_password"))) { > + fh->mm.auth_password = > switch_core_strdup(fh->memory_pool, val); > + } > + > if ((val = switch_event_get_header(fh->params, "fps"))) { > float ftmp = atof(val); > if (ftmp > 0.0f) { > (END) > > > I believe that it is the changes in switch_core_file.c that break my > utility program, but I can't be sure. Just o add some detail, I have a > Digium TCE400 card in my system which handles the G729 conversion. > > I have no idea of what those auth_username and auth_password parameters > are, but that seems to break things for me. Any idea of what I can do to > get things working again? > > I have created a PasteBin with the code of my fs_encode derived utility: > https://pastebin.freeswitch.org/view/a3d64c69 It is pretty short. > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at 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 ♬ @anthmfs ♬ @FreeSWITCH ♬ ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ * ClueCon Weekly Development Call ☎ sip:888 at conference.freeswitch.org ☎ +19193869900 https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA -------------- next part -------------- An HTML attachment was scrubbed... URL: From grcamauer at gmail.com Fri Jun 23 23:12:23 2017 From: grcamauer at gmail.com (Guillermo Ruiz Camauer) Date: Fri, 23 Jun 2017 20:12:23 -0300 Subject: [Freeswitch-users] Problem with fs_encode In-Reply-To: References: Message-ID: Michael, No core file is generated. Can I force one to be generated? How? Thanks, Guillermo On Fri, Jun 23, 2017 at 8:01 PM, Michael Jerris wrote: > can you drop a core file and see where its stuck? > > On Jun 23, 2017, at 6:57 PM, Guillermo Ruiz Camauer > wrote: > > I have created a small utility program based on fs_encode which will take > ALL files in a directory and convert them from .WAV to .G729 and .PCMU > formats. The utility is called wavBatchEncode and takes just a directory > path as an argument. > > This program was working very well until I upgraded FreeSwitch. Now the > program seems to hang when it tries to write out the first converted > files. I have to KILL the program from another terminal. It leaves 2 0 > byte files with .PCMU and .G729 extension. > > I have run through a GIT bisect to find where things broke. The last > working version is Version 1.6.16 git ae1cdce 2017-04-11. > The first broken version is 38621e47bad3b63f03a0a27f6ca9ed92f6969032. > > I then used GIT DIFF to see if I could see what had happened. There are > very few modifications between these two commits: > > root at fs3:/usr/src/freeswitch.git# git diff ae1cdce > 38621e47bad3b63f03a0a27f6ca9ed92f6969032 > diff --git a/src/include/switch_module_interfaces.h > b/src/include/switch_module_interfaces.h > index e0a5c20..7ca027d 100644 > --- a/src/include/switch_module_interfaces.h > +++ b/src/include/switch_module_interfaces.h > @@ -329,6 +329,11 @@ typedef struct switch_mm_s { > switch_video_profile_t vprofile; > switch_video_encode_speed_t vencspd; > uint8_t try_hardware_encoder; > + int scale_w; > + int scale_h; > + switch_img_fmt_t fmt; > + char *auth_username; > + char *auth_password; > } switch_mm_t; > > /*! an abstract representation of a file handle (some parameters based on > compat with libsndfile) */ > diff --git a/src/mod/applications/mod_av/avformat.c > b/src/mod/applications/mod_av/avformat.c > index b944625..4b92801 100644 > --- a/src/mod/applications/mod_av/avformat.c > +++ b/src/mod/applications/mod_av/avformat.c > @@ -906,7 +906,7 @@ SWITCH_STANDARD_APP(record_av_function) > char codec_str[256]; > const AVCodecDescriptor *desc; > > - if (!strncmp(data, "rtmp://", 7)) { > + if (!strncmp(data, "rtmp://", 7) || !strncmp(data, > "rtsp://", 7)) { > fmt->video_codec = AV_CODEC_ID_H264; > fmt->audio_codec = AV_CODEC_ID_AAC; > } > @@ -1694,9 +1694,20 @@ static switch_status_t av_file_open(switch_file_handle_t > *handle, const char *pa > return SWITCH_STATUS_GENERR; > } else if (handle->stream_name && (!strcasecmp(handle->stream_name, > "rtmp") || !strcasecmp(handle->stream_name, "youtube"))) { > format = "flv"; > - switch_snprintf(file, sizeof(file), "rtmp://%s", path); > + > + // meh really silly format for the user / pass libav..... > + if (handle->mm.auth_username && handle->mm.auth_password) > { > + switch_snprintf(file, sizeof(file), "rtmp://%s > pubUser=%s pubPasswd=%s flashver=FMLE/3.0", path, handle->mm.auth_username, > handle->mm.auth_password); > + } else { > + switch_snprintf(file, sizeof(file), "rtmp://%s", > path); > + } > + > + } else if (handle->stream_name && !strcasecmp(handle->stream_name, > "rtsp")) { > + format = "rtsp"; > + switch_snprintf(file, sizeof(file), "rtsp://%s", path); > } > > + > ext++; > > if ((context = (av_file_context_t *)switch_core_alloc(handle->memory_pool, > sizeof(av_file_context_t))) == 0) { > @@ -1783,7 +1794,7 @@ static switch_status_t av_file_open(switch_file_handle_t > *handle, const char *pa > if (fmt->video_codec != AV_CODEC_ID_NONE) { > const AVCodecDescriptor *desc; > > - if ((handle->stream_name && (!strcasecmp(handle->stream_name, > "rtmp") || !strcasecmp(handle->stream_name, "youtube")))) { > + if ((handle->stream_name && (!strcasecmp(handle->stream_name, > "rtmp") || !strcasecmp(handle->stream_name, "rtsp") || > !strcasecmp(handle->stream_name, "youtube")))) { > > if (fmt->video_codec != AV_CODEC_ID_H264 ) { > fmt->video_codec = AV_CODEC_ID_H264; // > force H264 > @@ -2525,6 +2536,7 @@ SWITCH_MODULE_LOAD_FUNCTION(mod_avformat_load) > > supported_formats[i++] = "av"; > supported_formats[i++] = "rtmp"; > + supported_formats[i++] = "rtsp"; > supported_formats[i++] = "mp4"; > supported_formats[i++] = "m4a"; > supported_formats[i++] = "mov"; > diff --git a/src/mod/applications/mod_av/mod_av.c > b/src/mod/applications/mod_av/mod_av.c > index 3d3bd82..141fcdc 100644 > --- a/src/mod/applications/mod_av/mod_av.c > +++ b/src/mod/applications/mod_av/mod_av.c > @@ -93,7 +93,7 @@ int mod_av_lockmgr_cb(void **m, enum AVLockOp op) > static void log_callback(void *ptr, int level, const char *fmt, va_list > vl) > { > switch_log_level_t switch_level = SWITCH_LOG_DEBUG; > - > + return; > /* naggy messages */ > if (level == AV_LOG_DEBUG || level == AV_LOG_WARNING) return; > > diff --git a/src/switch_core_file.c b/src/switch_core_file.c > index 46ee539..aff9442 100644 > --- a/src/switch_core_file.c > +++ b/src/switch_core_file.c > @@ -177,6 +177,14 @@ SWITCH_DECLARE(switch_status_t) > switch_core_perform_file_open(const char *file, > fh->mm.try_hardware_encoder = switch_true(val); > } > > + if ((val = switch_event_get_header(fh->params, > "auth_username"))) { > + fh->mm.auth_username = > switch_core_strdup(fh->memory_pool, val); > + } > + > + if ((val = switch_event_get_header(fh->params, > "auth_password"))) { > + fh->mm.auth_password = > switch_core_strdup(fh->memory_pool, val); > + } > + > if ((val = switch_event_get_header(fh->params, "fps"))) { > float ftmp = atof(val); > if (ftmp > 0.0f) { > (END) > > > I believe that it is the changes in switch_core_file.c that break my > utility program, but I can't be sure. Just o add some detail, I have a > Digium TCE400 card in my system which handles the G729 conversion. > > I have no idea of what those auth_username and auth_password parameters > are, but that seems to break things for me. Any idea of what I can do to > get things working again? > > I have created a PasteBin with the code of my fs_encode derived utility: > https://pastebin.freeswitch.org/view/a3d64c69 It is pretty short. > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Guillermo Ruiz Camauer -------------- next part -------------- An HTML attachment was scrubbed... URL: From grcamauer at gmail.com Fri Jun 23 23:14:04 2017 From: grcamauer at gmail.com (Guillermo Ruiz Camauer) Date: Fri, 23 Jun 2017 20:14:04 -0300 Subject: [Freeswitch-users] Problem with fs_encode In-Reply-To: References: Message-ID: I will try removing just those lines in switch_core_file.c. Maybe it's one of the other changes, I was just guessing... Guillermo On Fri, Jun 23, 2017 at 8:03 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > Also does removing just that change in switch_core_file.c from latest > version actually make it work? It seems unlikely the next line in the file > looking for FPS field is no different really. > > > > > On Fri, Jun 23, 2017 at 6:01 PM, Michael Jerris wrote: > >> can you drop a core file and see where its stuck? >> >> On Jun 23, 2017, at 6:57 PM, Guillermo Ruiz Camauer >> wrote: >> >> I have created a small utility program based on fs_encode which will take >> ALL files in a directory and convert them from .WAV to .G729 and .PCMU >> formats. The utility is called wavBatchEncode and takes just a directory >> path as an argument. >> >> This program was working very well until I upgraded FreeSwitch. Now the >> program seems to hang when it tries to write out the first converted >> files. I have to KILL the program from another terminal. It leaves 2 0 >> byte files with .PCMU and .G729 extension. >> >> I have run through a GIT bisect to find where things broke. The last >> working version is Version 1.6.16 git ae1cdce 2017-04-11. >> The first broken version is 38621e47bad3b63f03a0a27f6ca9ed92f6969032. >> >> I then used GIT DIFF to see if I could see what had happened. There are >> very few modifications between these two commits: >> >> root at fs3:/usr/src/freeswitch.git# git diff ae1cdce >> 38621e47bad3b63f03a0a27f6ca9ed92f6969032 >> diff --git a/src/include/switch_module_interfaces.h >> b/src/include/switch_module_interfaces.h >> index e0a5c20..7ca027d 100644 >> --- a/src/include/switch_module_interfaces.h >> +++ b/src/include/switch_module_interfaces.h >> @@ -329,6 +329,11 @@ typedef struct switch_mm_s { >> switch_video_profile_t vprofile; >> switch_video_encode_speed_t vencspd; >> uint8_t try_hardware_encoder; >> + int scale_w; >> + int scale_h; >> + switch_img_fmt_t fmt; >> + char *auth_username; >> + char *auth_password; >> } switch_mm_t; >> >> /*! an abstract representation of a file handle (some parameters based >> on compat with libsndfile) */ >> diff --git a/src/mod/applications/mod_av/avformat.c >> b/src/mod/applications/mod_av/avformat.c >> index b944625..4b92801 100644 >> --- a/src/mod/applications/mod_av/avformat.c >> +++ b/src/mod/applications/mod_av/avformat.c >> @@ -906,7 +906,7 @@ SWITCH_STANDARD_APP(record_av_function) >> char codec_str[256]; >> const AVCodecDescriptor *desc; >> >> - if (!strncmp(data, "rtmp://", 7)) { >> + if (!strncmp(data, "rtmp://", 7) || !strncmp(data, >> "rtsp://", 7)) { >> fmt->video_codec = AV_CODEC_ID_H264; >> fmt->audio_codec = AV_CODEC_ID_AAC; >> } >> @@ -1694,9 +1694,20 @@ static switch_status_t >> av_file_open(switch_file_handle_t *handle, const char *pa >> return SWITCH_STATUS_GENERR; >> } else if (handle->stream_name && (!strcasecmp(handle->stream_name, >> "rtmp") || !strcasecmp(handle->stream_name, "youtube"))) { >> format = "flv"; >> - switch_snprintf(file, sizeof(file), "rtmp://%s", path); >> + >> + // meh really silly format for the user / pass libav..... >> + if (handle->mm.auth_username && handle->mm.auth_password) >> { >> + switch_snprintf(file, sizeof(file), "rtmp://%s >> pubUser=%s pubPasswd=%s flashver=FMLE/3.0", path, handle->mm.auth_username, >> handle->mm.auth_password); >> + } else { >> + switch_snprintf(file, sizeof(file), "rtmp://%s", >> path); >> + } >> + >> + } else if (handle->stream_name && !strcasecmp(handle->stream_name, >> "rtsp")) { >> + format = "rtsp"; >> + switch_snprintf(file, sizeof(file), "rtsp://%s", path); >> } >> >> + >> ext++; >> >> if ((context = (av_file_context_t *)switch_core_alloc(handle->memory_pool, >> sizeof(av_file_context_t))) == 0) { >> @@ -1783,7 +1794,7 @@ static switch_status_t >> av_file_open(switch_file_handle_t *handle, const char *pa >> if (fmt->video_codec != AV_CODEC_ID_NONE) { >> const AVCodecDescriptor *desc; >> >> - if ((handle->stream_name && (!strcasecmp(handle->stream_name, >> "rtmp") || !strcasecmp(handle->stream_name, "youtube")))) { >> + if ((handle->stream_name && (!strcasecmp(handle->stream_name, >> "rtmp") || !strcasecmp(handle->stream_name, "rtsp") || >> !strcasecmp(handle->stream_name, "youtube")))) { >> >> if (fmt->video_codec != AV_CODEC_ID_H264 ) { >> fmt->video_codec = AV_CODEC_ID_H264; // >> force H264 >> @@ -2525,6 +2536,7 @@ SWITCH_MODULE_LOAD_FUNCTION(mod_avformat_load) >> >> supported_formats[i++] = "av"; >> supported_formats[i++] = "rtmp"; >> + supported_formats[i++] = "rtsp"; >> supported_formats[i++] = "mp4"; >> supported_formats[i++] = "m4a"; >> supported_formats[i++] = "mov"; >> diff --git a/src/mod/applications/mod_av/mod_av.c >> b/src/mod/applications/mod_av/mod_av.c >> index 3d3bd82..141fcdc 100644 >> --- a/src/mod/applications/mod_av/mod_av.c >> +++ b/src/mod/applications/mod_av/mod_av.c >> @@ -93,7 +93,7 @@ int mod_av_lockmgr_cb(void **m, enum AVLockOp op) >> static void log_callback(void *ptr, int level, const char *fmt, va_list >> vl) >> { >> switch_log_level_t switch_level = SWITCH_LOG_DEBUG; >> - >> + return; >> /* naggy messages */ >> if (level == AV_LOG_DEBUG || level == AV_LOG_WARNING) return; >> >> diff --git a/src/switch_core_file.c b/src/switch_core_file.c >> index 46ee539..aff9442 100644 >> --- a/src/switch_core_file.c >> +++ b/src/switch_core_file.c >> @@ -177,6 +177,14 @@ SWITCH_DECLARE(switch_status_t) >> switch_core_perform_file_open(const char *file, >> fh->mm.try_hardware_encoder = switch_true(val); >> } >> >> + if ((val = switch_event_get_header(fh->params, >> "auth_username"))) { >> + fh->mm.auth_username = >> switch_core_strdup(fh->memory_pool, val); >> + } >> + >> + if ((val = switch_event_get_header(fh->params, >> "auth_password"))) { >> + fh->mm.auth_password = >> switch_core_strdup(fh->memory_pool, val); >> + } >> + >> if ((val = switch_event_get_header(fh->params, "fps"))) { >> float ftmp = atof(val); >> if (ftmp > 0.0f) { >> (END) >> >> >> I believe that it is the changes in switch_core_file.c that break my >> utility program, but I can't be sure. Just o add some detail, I have a >> Digium TCE400 card in my system which handles the G729 conversion. >> >> I have no idea of what those auth_username and auth_password parameters >> are, but that seems to break things for me. Any idea of what I can do to >> get things working again? >> >> I have created a PasteBin with the code of my fs_encode derived utility: >> https://pastebin.freeswitch.org/view/a3d64c69 It is pretty short. >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at 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 ♬ @anthmfs ♬ @FreeSWITCH ♬ > > ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ > http://twitter.com/FreeSWITCH > ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ > * > > ClueCon Weekly Development Call > ☎ sip:888 at conference.freeswitch.org ☎ +19193869900 > > https://www.youtube.com/watch?v=9XXgW34t40s > https://www.youtube.com/watch?v=NLaDpGQuZDA > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Guillermo Ruiz Camauer -------------- next part -------------- An HTML attachment was scrubbed... URL: From anthony.minessale at gmail.com Fri Jun 23 23:16:38 2017 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 23 Jun 2017 18:16:38 -0500 Subject: [Freeswitch-users] Problem with fs_encode In-Reply-To: References: Message-ID: My advise is to verify by removing bits or even all of the patch and double check you did not mess up on the bisect. like if you can manually reproduce going to the version before this patch and having it work then go to the version with the patch and try again etc. On Fri, Jun 23, 2017 at 6:14 PM, Guillermo Ruiz Camauer wrote: > I will try removing just those lines in switch_core_file.c. Maybe it's > one of the other changes, I was just guessing... > > Guillermo > > On Fri, Jun 23, 2017 at 8:03 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> Also does removing just that change in switch_core_file.c from latest >> version actually make it work? It seems unlikely the next line in the file >> looking for FPS field is no different really. >> >> >> >> >> On Fri, Jun 23, 2017 at 6:01 PM, Michael Jerris wrote: >> >>> can you drop a core file and see where its stuck? >>> >>> On Jun 23, 2017, at 6:57 PM, Guillermo Ruiz Camauer >>> wrote: >>> >>> I have created a small utility program based on fs_encode which will >>> take ALL files in a directory and convert them from .WAV to .G729 and .PCMU >>> formats. The utility is called wavBatchEncode and takes just a directory >>> path as an argument. >>> >>> This program was working very well until I upgraded FreeSwitch. Now the >>> program seems to hang when it tries to write out the first converted >>> files. I have to KILL the program from another terminal. It leaves 2 0 >>> byte files with .PCMU and .G729 extension. >>> >>> I have run through a GIT bisect to find where things broke. The last >>> working version is Version 1.6.16 git ae1cdce 2017-04-11. >>> The first broken version is 38621e47bad3b63f03a0a27f6ca9ed92f6969032. >>> >>> I then used GIT DIFF to see if I could see what had happened. There are >>> very few modifications between these two commits: >>> >>> root at fs3:/usr/src/freeswitch.git# git diff ae1cdce >>> 38621e47bad3b63f03a0a27f6ca9ed92f6969032 >>> diff --git a/src/include/switch_module_interfaces.h >>> b/src/include/switch_module_interfaces.h >>> index e0a5c20..7ca027d 100644 >>> --- a/src/include/switch_module_interfaces.h >>> +++ b/src/include/switch_module_interfaces.h >>> @@ -329,6 +329,11 @@ typedef struct switch_mm_s { >>> switch_video_profile_t vprofile; >>> switch_video_encode_speed_t vencspd; >>> uint8_t try_hardware_encoder; >>> + int scale_w; >>> + int scale_h; >>> + switch_img_fmt_t fmt; >>> + char *auth_username; >>> + char *auth_password; >>> } switch_mm_t; >>> >>> /*! an abstract representation of a file handle (some parameters based >>> on compat with libsndfile) */ >>> diff --git a/src/mod/applications/mod_av/avformat.c >>> b/src/mod/applications/mod_av/avformat.c >>> index b944625..4b92801 100644 >>> --- a/src/mod/applications/mod_av/avformat.c >>> +++ b/src/mod/applications/mod_av/avformat.c >>> @@ -906,7 +906,7 @@ SWITCH_STANDARD_APP(record_av_function) >>> char codec_str[256]; >>> const AVCodecDescriptor *desc; >>> >>> - if (!strncmp(data, "rtmp://", 7)) { >>> + if (!strncmp(data, "rtmp://", 7) || !strncmp(data, >>> "rtsp://", 7)) { >>> fmt->video_codec = AV_CODEC_ID_H264; >>> fmt->audio_codec = AV_CODEC_ID_AAC; >>> } >>> @@ -1694,9 +1694,20 @@ static switch_status_t >>> av_file_open(switch_file_handle_t *handle, const char *pa >>> return SWITCH_STATUS_GENERR; >>> } else if (handle->stream_name && (!strcasecmp(handle->stream_name, >>> "rtmp") || !strcasecmp(handle->stream_name, "youtube"))) { >>> format = "flv"; >>> - switch_snprintf(file, sizeof(file), "rtmp://%s", path); >>> + >>> + // meh really silly format for the user / pass libav..... >>> + if (handle->mm.auth_username && >>> handle->mm.auth_password) { >>> + switch_snprintf(file, sizeof(file), "rtmp://%s >>> pubUser=%s pubPasswd=%s flashver=FMLE/3.0", path, handle->mm.auth_username, >>> handle->mm.auth_password); >>> + } else { >>> + switch_snprintf(file, sizeof(file), "rtmp://%s", >>> path); >>> + } >>> + >>> + } else if (handle->stream_name && !strcasecmp(handle->stream_name, >>> "rtsp")) { >>> + format = "rtsp"; >>> + switch_snprintf(file, sizeof(file), "rtsp://%s", path); >>> } >>> >>> + >>> ext++; >>> >>> if ((context = (av_file_context_t *)switch_core_alloc(handle->memory_pool, >>> sizeof(av_file_context_t))) == 0) { >>> @@ -1783,7 +1794,7 @@ static switch_status_t >>> av_file_open(switch_file_handle_t *handle, const char *pa >>> if (fmt->video_codec != AV_CODEC_ID_NONE) { >>> const AVCodecDescriptor *desc; >>> >>> - if ((handle->stream_name && >>> (!strcasecmp(handle->stream_name, "rtmp") || >>> !strcasecmp(handle->stream_name, "youtube")))) { >>> + if ((handle->stream_name && >>> (!strcasecmp(handle->stream_name, "rtmp") || >>> !strcasecmp(handle->stream_name, "rtsp") || >>> !strcasecmp(handle->stream_name, "youtube")))) { >>> >>> if (fmt->video_codec != AV_CODEC_ID_H264 ) { >>> fmt->video_codec = AV_CODEC_ID_H264; // >>> force H264 >>> @@ -2525,6 +2536,7 @@ SWITCH_MODULE_LOAD_FUNCTION(mod_avformat_load) >>> >>> supported_formats[i++] = "av"; >>> supported_formats[i++] = "rtmp"; >>> + supported_formats[i++] = "rtsp"; >>> supported_formats[i++] = "mp4"; >>> supported_formats[i++] = "m4a"; >>> supported_formats[i++] = "mov"; >>> diff --git a/src/mod/applications/mod_av/mod_av.c >>> b/src/mod/applications/mod_av/mod_av.c >>> index 3d3bd82..141fcdc 100644 >>> --- a/src/mod/applications/mod_av/mod_av.c >>> +++ b/src/mod/applications/mod_av/mod_av.c >>> @@ -93,7 +93,7 @@ int mod_av_lockmgr_cb(void **m, enum AVLockOp op) >>> static void log_callback(void *ptr, int level, const char *fmt, va_list >>> vl) >>> { >>> switch_log_level_t switch_level = SWITCH_LOG_DEBUG; >>> - >>> + return; >>> /* naggy messages */ >>> if (level == AV_LOG_DEBUG || level == AV_LOG_WARNING) return; >>> >>> diff --git a/src/switch_core_file.c b/src/switch_core_file.c >>> index 46ee539..aff9442 100644 >>> --- a/src/switch_core_file.c >>> +++ b/src/switch_core_file.c >>> @@ -177,6 +177,14 @@ SWITCH_DECLARE(switch_status_t) >>> switch_core_perform_file_open(const char *file, >>> fh->mm.try_hardware_encoder = switch_true(val); >>> } >>> >>> + if ((val = switch_event_get_header(fh->params, >>> "auth_username"))) { >>> + fh->mm.auth_username = >>> switch_core_strdup(fh->memory_pool, val); >>> + } >>> + >>> + if ((val = switch_event_get_header(fh->params, >>> "auth_password"))) { >>> + fh->mm.auth_password = >>> switch_core_strdup(fh->memory_pool, val); >>> + } >>> + >>> if ((val = switch_event_get_header(fh->params, "fps"))) >>> { >>> float ftmp = atof(val); >>> if (ftmp > 0.0f) { >>> (END) >>> >>> >>> I believe that it is the changes in switch_core_file.c that break my >>> utility program, but I can't be sure. Just o add some detail, I have a >>> Digium TCE400 card in my system which handles the G729 conversion. >>> >>> I have no idea of what those auth_username and auth_password parameters >>> are, but that seems to break things for me. Any idea of what I can do to >>> get things working again? >>> >>> I have created a PasteBin with the code of my fs_encode derived utility: >>> https://pastebin.freeswitch.org/view/a3d64c69 It is pretty short. >>> >>> >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at 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 at 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 ♬ @anthmfs ♬ @FreeSWITCH ♬ >> >> ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ >> http://twitter.com/FreeSWITCH >> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ >> * >> >> ClueCon Weekly Development Call >> ☎ sip:888 at conference.freeswitch.org ☎ +19193869900 <(919)%20386-9900> >> >> https://www.youtube.com/watch?v=9XXgW34t40s >> https://www.youtube.com/watch?v=NLaDpGQuZDA >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Guillermo Ruiz Camauer > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at 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 ♬ @anthmfs ♬ @FreeSWITCH ♬ ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ * ClueCon Weekly Development Call ☎ sip:888 at conference.freeswitch.org ☎ +19193869900 https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA -------------- next part -------------- An HTML attachment was scrubbed... URL: From grcamauer at gmail.com Fri Jun 23 23:24:06 2017 From: grcamauer at gmail.com (Guillermo Ruiz Camauer) Date: Fri, 23 Jun 2017 20:24:06 -0300 Subject: [Freeswitch-users] Problem with fs_encode In-Reply-To: References: Message-ID: Anthony, Thaks for your quick response! I have verified that it works with the previous commit (ae1cdce). Commit 38621e47bad3b63f03a0a27f6ca9ed92f6969032 is the first one that breaks it. I just removed the following lines from switch_core_file.c and recompiled: + if ((val = switch_event_get_header(fh->params, "auth_username"))) { + fh->mm.auth_username = switch_core_strdup(fh->memory_pool, val); + } + + if ((val = switch_event_get_header(fh->params, "auth_password"))) { + fh->mm.auth_password = switch_core_strdup(fh->memory_pool, val); + } Still broken... I will try removing other bits to see what happens. Guillermo On Fri, Jun 23, 2017 at 8:16 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > My advise is to verify by removing bits or even all of the patch and > double check you did not mess up on the bisect. > like if you can manually reproduce going to the version before this patch > and having it work then go to the version with the patch and try again etc. > > > > On Fri, Jun 23, 2017 at 6:14 PM, Guillermo Ruiz Camauer < > grcamauer at gmail.com> wrote: > >> I will try removing just those lines in switch_core_file.c. Maybe it's >> one of the other changes, I was just guessing... >> >> Guillermo >> >> On Fri, Jun 23, 2017 at 8:03 PM, Anthony Minessale < >> anthony.minessale at gmail.com> wrote: >> >>> Also does removing just that change in switch_core_file.c from latest >>> version actually make it work? It seems unlikely the next line in the file >>> looking for FPS field is no different really. >>> >>> >>> >>> >>> On Fri, Jun 23, 2017 at 6:01 PM, Michael Jerris wrote: >>> >>>> can you drop a core file and see where its stuck? >>>> >>>> On Jun 23, 2017, at 6:57 PM, Guillermo Ruiz Camauer < >>>> grcamauer at gmail.com> wrote: >>>> >>>> I have created a small utility program based on fs_encode which will >>>> take ALL files in a directory and convert them from .WAV to .G729 and .PCMU >>>> formats. The utility is called wavBatchEncode and takes just a directory >>>> path as an argument. >>>> >>>> This program was working very well until I upgraded FreeSwitch. Now the >>>> program seems to hang when it tries to write out the first converted >>>> files. I have to KILL the program from another terminal. It leaves 2 0 >>>> byte files with .PCMU and .G729 extension. >>>> >>>> I have run through a GIT bisect to find where things broke. The last >>>> working version is Version 1.6.16 git ae1cdce 2017-04-11. >>>> The first broken version is 38621e47bad3b63f03a0a27f6ca9ed92f6969032. >>>> >>>> I then used GIT DIFF to see if I could see what had happened. There >>>> are very few modifications between these two commits: >>>> >>>> root at fs3:/usr/src/freeswitch.git# git diff ae1cdce >>>> 38621e47bad3b63f03a0a27f6ca9ed92f6969032 >>>> diff --git a/src/include/switch_module_interfaces.h >>>> b/src/include/switch_module_interfaces.h >>>> index e0a5c20..7ca027d 100644 >>>> --- a/src/include/switch_module_interfaces.h >>>> +++ b/src/include/switch_module_interfaces.h >>>> @@ -329,6 +329,11 @@ typedef struct switch_mm_s { >>>> switch_video_profile_t vprofile; >>>> switch_video_encode_speed_t vencspd; >>>> uint8_t try_hardware_encoder; >>>> + int scale_w; >>>> + int scale_h; >>>> + switch_img_fmt_t fmt; >>>> + char *auth_username; >>>> + char *auth_password; >>>> } switch_mm_t; >>>> >>>> /*! an abstract representation of a file handle (some parameters based >>>> on compat with libsndfile) */ >>>> diff --git a/src/mod/applications/mod_av/avformat.c >>>> b/src/mod/applications/mod_av/avformat.c >>>> index b944625..4b92801 100644 >>>> --- a/src/mod/applications/mod_av/avformat.c >>>> +++ b/src/mod/applications/mod_av/avformat.c >>>> @@ -906,7 +906,7 @@ SWITCH_STANDARD_APP(record_av_function) >>>> char codec_str[256]; >>>> const AVCodecDescriptor *desc; >>>> >>>> - if (!strncmp(data, "rtmp://", 7)) { >>>> + if (!strncmp(data, "rtmp://", 7) || !strncmp(data, >>>> "rtsp://", 7)) { >>>> fmt->video_codec = AV_CODEC_ID_H264; >>>> fmt->audio_codec = AV_CODEC_ID_AAC; >>>> } >>>> @@ -1694,9 +1694,20 @@ static switch_status_t >>>> av_file_open(switch_file_handle_t *handle, const char *pa >>>> return SWITCH_STATUS_GENERR; >>>> } else if (handle->stream_name && (!strcasecmp(handle->stream_name, >>>> "rtmp") || !strcasecmp(handle->stream_name, "youtube"))) { >>>> format = "flv"; >>>> - switch_snprintf(file, sizeof(file), "rtmp://%s", path); >>>> + >>>> + // meh really silly format for the user / pass >>>> libav..... >>>> + if (handle->mm.auth_username && >>>> handle->mm.auth_password) { >>>> + switch_snprintf(file, sizeof(file), "rtmp://%s >>>> pubUser=%s pubPasswd=%s flashver=FMLE/3.0", path, handle->mm.auth_username, >>>> handle->mm.auth_password); >>>> + } else { >>>> + switch_snprintf(file, sizeof(file), "rtmp://%s", >>>> path); >>>> + } >>>> + >>>> + } else if (handle->stream_name && !strcasecmp(handle->stream_name, >>>> "rtsp")) { >>>> + format = "rtsp"; >>>> + switch_snprintf(file, sizeof(file), "rtsp://%s", path); >>>> } >>>> >>>> + >>>> ext++; >>>> >>>> if ((context = (av_file_context_t *)switch_core_alloc(handle->memory_pool, >>>> sizeof(av_file_context_t))) == 0) { >>>> @@ -1783,7 +1794,7 @@ static switch_status_t >>>> av_file_open(switch_file_handle_t *handle, const char *pa >>>> if (fmt->video_codec != AV_CODEC_ID_NONE) { >>>> const AVCodecDescriptor *desc; >>>> >>>> - if ((handle->stream_name && >>>> (!strcasecmp(handle->stream_name, "rtmp") || >>>> !strcasecmp(handle->stream_name, "youtube")))) { >>>> + if ((handle->stream_name && >>>> (!strcasecmp(handle->stream_name, "rtmp") || >>>> !strcasecmp(handle->stream_name, "rtsp") || >>>> !strcasecmp(handle->stream_name, "youtube")))) { >>>> >>>> if (fmt->video_codec != AV_CODEC_ID_H264 ) { >>>> fmt->video_codec = AV_CODEC_ID_H264; // >>>> force H264 >>>> @@ -2525,6 +2536,7 @@ SWITCH_MODULE_LOAD_FUNCTION(mod_avformat_load) >>>> >>>> supported_formats[i++] = "av"; >>>> supported_formats[i++] = "rtmp"; >>>> + supported_formats[i++] = "rtsp"; >>>> supported_formats[i++] = "mp4"; >>>> supported_formats[i++] = "m4a"; >>>> supported_formats[i++] = "mov"; >>>> diff --git a/src/mod/applications/mod_av/mod_av.c >>>> b/src/mod/applications/mod_av/mod_av.c >>>> index 3d3bd82..141fcdc 100644 >>>> --- a/src/mod/applications/mod_av/mod_av.c >>>> +++ b/src/mod/applications/mod_av/mod_av.c >>>> @@ -93,7 +93,7 @@ int mod_av_lockmgr_cb(void **m, enum AVLockOp op) >>>> static void log_callback(void *ptr, int level, const char *fmt, >>>> va_list vl) >>>> { >>>> switch_log_level_t switch_level = SWITCH_LOG_DEBUG; >>>> - >>>> + return; >>>> /* naggy messages */ >>>> if (level == AV_LOG_DEBUG || level == AV_LOG_WARNING) return; >>>> >>>> diff --git a/src/switch_core_file.c b/src/switch_core_file.c >>>> index 46ee539..aff9442 100644 >>>> --- a/src/switch_core_file.c >>>> +++ b/src/switch_core_file.c >>>> @@ -177,6 +177,14 @@ SWITCH_DECLARE(switch_status_t) >>>> switch_core_perform_file_open(const char *file, >>>> fh->mm.try_hardware_encoder = switch_true(val); >>>> } >>>> >>>> + if ((val = switch_event_get_header(fh->params, >>>> "auth_username"))) { >>>> + fh->mm.auth_username = >>>> switch_core_strdup(fh->memory_pool, val); >>>> + } >>>> + >>>> + if ((val = switch_event_get_header(fh->params, >>>> "auth_password"))) { >>>> + fh->mm.auth_password = >>>> switch_core_strdup(fh->memory_pool, val); >>>> + } >>>> + >>>> if ((val = switch_event_get_header(fh->params, >>>> "fps"))) { >>>> float ftmp = atof(val); >>>> if (ftmp > 0.0f) { >>>> (END) >>>> >>>> >>>> I believe that it is the changes in switch_core_file.c that break my >>>> utility program, but I can't be sure. Just o add some detail, I have a >>>> Digium TCE400 card in my system which handles the G729 conversion. >>>> >>>> I have no idea of what those auth_username and auth_password parameters >>>> are, but that seems to break things for me. Any idea of what I can do to >>>> get things working again? >>>> >>>> I have created a PasteBin with the code of my fs_encode derived >>>> utility: https://pastebin.freeswitch.org/view/a3d64c69 It is pretty >>>> short. >>>> >>>> >>>> >>>> ____________________________________________________________ >>>> _____________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at 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 at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>> switch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> >>> -- >>> Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ >>> >>> ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ >>> http://twitter.com/FreeSWITCH >>> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ >>> * >>> >>> ClueCon Weekly Development Call >>> ☎ sip:888 at conference.freeswitch.org ☎ +19193869900 <(919)%20386-9900> >>> >>> https://www.youtube.com/watch?v=9XXgW34t40s >>> https://www.youtube.com/watch?v=NLaDpGQuZDA >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at 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 at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Guillermo Ruiz Camauer >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at 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 ♬ @anthmfs ♬ @FreeSWITCH ♬ > > ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ > http://twitter.com/FreeSWITCH > ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ > * > > ClueCon Weekly Development Call > ☎ sip:888 at conference.freeswitch.org ☎ +19193869900 > > https://www.youtube.com/watch?v=9XXgW34t40s > https://www.youtube.com/watch?v=NLaDpGQuZDA > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Guillermo Ruiz Camauer -------------- next part -------------- An HTML attachment was scrubbed... URL: From grcamauer at gmail.com Fri Jun 23 23:34:19 2017 From: grcamauer at gmail.com (Guillermo Ruiz Camauer) Date: Fri, 23 Jun 2017 20:34:19 -0300 Subject: [Freeswitch-users] Problem with fs_encode In-Reply-To: References: Message-ID: Anthony, Removing these lines: + int scale_w; + int scale_h; + switch_img_fmt_t fmt; + char *auth_username; + char *auth_password; from switch_module_interfaces.h makes my program work again. Any idea of what they could be affecting? Guillermo On Fri, Jun 23, 2017 at 8:16 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > My advise is to verify by removing bits or even all of the patch and > double check you did not mess up on the bisect. > like if you can manually reproduce going to the version before this patch > and having it work then go to the version with the patch and try again etc. > > > > On Fri, Jun 23, 2017 at 6:14 PM, Guillermo Ruiz Camauer < > grcamauer at gmail.com> wrote: > >> I will try removing just those lines in switch_core_file.c. Maybe it's >> one of the other changes, I was just guessing... >> >> Guillermo >> >> On Fri, Jun 23, 2017 at 8:03 PM, Anthony Minessale < >> anthony.minessale at gmail.com> wrote: >> >>> Also does removing just that change in switch_core_file.c from latest >>> version actually make it work? It seems unlikely the next line in the file >>> looking for FPS field is no different really. >>> >>> >>> >>> >>> On Fri, Jun 23, 2017 at 6:01 PM, Michael Jerris wrote: >>> >>>> can you drop a core file and see where its stuck? >>>> >>>> On Jun 23, 2017, at 6:57 PM, Guillermo Ruiz Camauer < >>>> grcamauer at gmail.com> wrote: >>>> >>>> I have created a small utility program based on fs_encode which will >>>> take ALL files in a directory and convert them from .WAV to .G729 and .PCMU >>>> formats. The utility is called wavBatchEncode and takes just a directory >>>> path as an argument. >>>> >>>> This program was working very well until I upgraded FreeSwitch. Now the >>>> program seems to hang when it tries to write out the first converted >>>> files. I have to KILL the program from another terminal. It leaves 2 0 >>>> byte files with .PCMU and .G729 extension. >>>> >>>> I have run through a GIT bisect to find where things broke. The last >>>> working version is Version 1.6.16 git ae1cdce 2017-04-11. >>>> The first broken version is 38621e47bad3b63f03a0a27f6ca9ed92f6969032. >>>> >>>> I then used GIT DIFF to see if I could see what had happened. There >>>> are very few modifications between these two commits: >>>> >>>> root at fs3:/usr/src/freeswitch.git# git diff ae1cdce >>>> 38621e47bad3b63f03a0a27f6ca9ed92f6969032 >>>> diff --git a/src/include/switch_module_interfaces.h >>>> b/src/include/switch_module_interfaces.h >>>> index e0a5c20..7ca027d 100644 >>>> --- a/src/include/switch_module_interfaces.h >>>> +++ b/src/include/switch_module_interfaces.h >>>> @@ -329,6 +329,11 @@ typedef struct switch_mm_s { >>>> switch_video_profile_t vprofile; >>>> switch_video_encode_speed_t vencspd; >>>> uint8_t try_hardware_encoder; >>>> + int scale_w; >>>> + int scale_h; >>>> + switch_img_fmt_t fmt; >>>> + char *auth_username; >>>> + char *auth_password; >>>> } switch_mm_t; >>>> >>>> /*! an abstract representation of a file handle (some parameters based >>>> on compat with libsndfile) */ >>>> diff --git a/src/mod/applications/mod_av/avformat.c >>>> b/src/mod/applications/mod_av/avformat.c >>>> index b944625..4b92801 100644 >>>> --- a/src/mod/applications/mod_av/avformat.c >>>> +++ b/src/mod/applications/mod_av/avformat.c >>>> @@ -906,7 +906,7 @@ SWITCH_STANDARD_APP(record_av_function) >>>> char codec_str[256]; >>>> const AVCodecDescriptor *desc; >>>> >>>> - if (!strncmp(data, "rtmp://", 7)) { >>>> + if (!strncmp(data, "rtmp://", 7) || !strncmp(data, >>>> "rtsp://", 7)) { >>>> fmt->video_codec = AV_CODEC_ID_H264; >>>> fmt->audio_codec = AV_CODEC_ID_AAC; >>>> } >>>> @@ -1694,9 +1694,20 @@ static switch_status_t >>>> av_file_open(switch_file_handle_t *handle, const char *pa >>>> return SWITCH_STATUS_GENERR; >>>> } else if (handle->stream_name && (!strcasecmp(handle->stream_name, >>>> "rtmp") || !strcasecmp(handle->stream_name, "youtube"))) { >>>> format = "flv"; >>>> - switch_snprintf(file, sizeof(file), "rtmp://%s", path); >>>> + >>>> + // meh really silly format for the user / pass >>>> libav..... >>>> + if (handle->mm.auth_username && >>>> handle->mm.auth_password) { >>>> + switch_snprintf(file, sizeof(file), "rtmp://%s >>>> pubUser=%s pubPasswd=%s flashver=FMLE/3.0", path, handle->mm.auth_username, >>>> handle->mm.auth_password); >>>> + } else { >>>> + switch_snprintf(file, sizeof(file), "rtmp://%s", >>>> path); >>>> + } >>>> + >>>> + } else if (handle->stream_name && !strcasecmp(handle->stream_name, >>>> "rtsp")) { >>>> + format = "rtsp"; >>>> + switch_snprintf(file, sizeof(file), "rtsp://%s", path); >>>> } >>>> >>>> + >>>> ext++; >>>> >>>> if ((context = (av_file_context_t *)switch_core_alloc(handle->memory_pool, >>>> sizeof(av_file_context_t))) == 0) { >>>> @@ -1783,7 +1794,7 @@ static switch_status_t >>>> av_file_open(switch_file_handle_t *handle, const char *pa >>>> if (fmt->video_codec != AV_CODEC_ID_NONE) { >>>> const AVCodecDescriptor *desc; >>>> >>>> - if ((handle->stream_name && >>>> (!strcasecmp(handle->stream_name, "rtmp") || >>>> !strcasecmp(handle->stream_name, "youtube")))) { >>>> + if ((handle->stream_name && >>>> (!strcasecmp(handle->stream_name, "rtmp") || >>>> !strcasecmp(handle->stream_name, "rtsp") || >>>> !strcasecmp(handle->stream_name, "youtube")))) { >>>> >>>> if (fmt->video_codec != AV_CODEC_ID_H264 ) { >>>> fmt->video_codec = AV_CODEC_ID_H264; // >>>> force H264 >>>> @@ -2525,6 +2536,7 @@ SWITCH_MODULE_LOAD_FUNCTION(mod_avformat_load) >>>> >>>> supported_formats[i++] = "av"; >>>> supported_formats[i++] = "rtmp"; >>>> + supported_formats[i++] = "rtsp"; >>>> supported_formats[i++] = "mp4"; >>>> supported_formats[i++] = "m4a"; >>>> supported_formats[i++] = "mov"; >>>> diff --git a/src/mod/applications/mod_av/mod_av.c >>>> b/src/mod/applications/mod_av/mod_av.c >>>> index 3d3bd82..141fcdc 100644 >>>> --- a/src/mod/applications/mod_av/mod_av.c >>>> +++ b/src/mod/applications/mod_av/mod_av.c >>>> @@ -93,7 +93,7 @@ int mod_av_lockmgr_cb(void **m, enum AVLockOp op) >>>> static void log_callback(void *ptr, int level, const char *fmt, >>>> va_list vl) >>>> { >>>> switch_log_level_t switch_level = SWITCH_LOG_DEBUG; >>>> - >>>> + return; >>>> /* naggy messages */ >>>> if (level == AV_LOG_DEBUG || level == AV_LOG_WARNING) return; >>>> >>>> diff --git a/src/switch_core_file.c b/src/switch_core_file.c >>>> index 46ee539..aff9442 100644 >>>> --- a/src/switch_core_file.c >>>> +++ b/src/switch_core_file.c >>>> @@ -177,6 +177,14 @@ SWITCH_DECLARE(switch_status_t) >>>> switch_core_perform_file_open(const char *file, >>>> fh->mm.try_hardware_encoder = switch_true(val); >>>> } >>>> >>>> + if ((val = switch_event_get_header(fh->params, >>>> "auth_username"))) { >>>> + fh->mm.auth_username = >>>> switch_core_strdup(fh->memory_pool, val); >>>> + } >>>> + >>>> + if ((val = switch_event_get_header(fh->params, >>>> "auth_password"))) { >>>> + fh->mm.auth_password = >>>> switch_core_strdup(fh->memory_pool, val); >>>> + } >>>> + >>>> if ((val = switch_event_get_header(fh->params, >>>> "fps"))) { >>>> float ftmp = atof(val); >>>> if (ftmp > 0.0f) { >>>> (END) >>>> >>>> >>>> I believe that it is the changes in switch_core_file.c that break my >>>> utility program, but I can't be sure. Just o add some detail, I have a >>>> Digium TCE400 card in my system which handles the G729 conversion. >>>> >>>> I have no idea of what those auth_username and auth_password parameters >>>> are, but that seems to break things for me. Any idea of what I can do to >>>> get things working again? >>>> >>>> I have created a PasteBin with the code of my fs_encode derived >>>> utility: https://pastebin.freeswitch.org/view/a3d64c69 It is pretty >>>> short. >>>> >>>> >>>> >>>> ____________________________________________________________ >>>> _____________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at 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 at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>> switch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> >>> -- >>> Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ >>> >>> ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ >>> http://twitter.com/FreeSWITCH >>> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ >>> * >>> >>> ClueCon Weekly Development Call >>> ☎ sip:888 at conference.freeswitch.org ☎ +19193869900 <(919)%20386-9900> >>> >>> https://www.youtube.com/watch?v=9XXgW34t40s >>> https://www.youtube.com/watch?v=NLaDpGQuZDA >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at 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 at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Guillermo Ruiz Camauer >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at 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 ♬ @anthmfs ♬ @FreeSWITCH ♬ > > ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ > http://twitter.com/FreeSWITCH > ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ > * > > ClueCon Weekly Development Call > ☎ sip:888 at conference.freeswitch.org ☎ +19193869900 > > https://www.youtube.com/watch?v=9XXgW34t40s > https://www.youtube.com/watch?v=NLaDpGQuZDA > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Guillermo Ruiz Camauer -------------- next part -------------- An HTML attachment was scrubbed... URL: From anthony.minessale at gmail.com Sat Jun 24 03:57:11 2017 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 24 Jun 2017 03:57:11 +0000 Subject: [Freeswitch-users] Problem with fs_encode In-Reply-To: References: Message-ID: Are you recompiling the app too against the new code. The patch changes abi of the fs libs. On Fri, Jun 23, 2017 at 6:35 PM Guillermo Ruiz Camauer wrote: > Anthony, > > > Removing these lines: > > + int scale_w; > + int scale_h; > + switch_img_fmt_t fmt; > + char *auth_username; > + char *auth_password; > > from switch_module_interfaces.h > > makes my program work again. Any idea of what they could be affecting? > > Guillermo > > On Fri, Jun 23, 2017 at 8:16 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> My advise is to verify by removing bits or even all of the patch and >> double check you did not mess up on the bisect. >> like if you can manually reproduce going to the version before this patch >> and having it work then go to the version with the patch and try again etc. >> >> >> >> On Fri, Jun 23, 2017 at 6:14 PM, Guillermo Ruiz Camauer < >> grcamauer at gmail.com> wrote: >> >>> I will try removing just those lines in switch_core_file.c. Maybe it's >>> one of the other changes, I was just guessing... >>> >>> Guillermo >>> >>> On Fri, Jun 23, 2017 at 8:03 PM, Anthony Minessale < >>> anthony.minessale at gmail.com> wrote: >>> >>>> Also does removing just that change in switch_core_file.c from latest >>>> version actually make it work? It seems unlikely the next line in the file >>>> looking for FPS field is no different really. >>>> >>>> >>>> >>>> >>>> On Fri, Jun 23, 2017 at 6:01 PM, Michael Jerris >>>> wrote: >>>> >>>>> can you drop a core file and see where its stuck? >>>>> >>>>> On Jun 23, 2017, at 6:57 PM, Guillermo Ruiz Camauer < >>>>> grcamauer at gmail.com> wrote: >>>>> >>>>> I have created a small utility program based on fs_encode which will >>>>> take ALL files in a directory and convert them from .WAV to .G729 and .PCMU >>>>> formats. The utility is called wavBatchEncode and takes just a directory >>>>> path as an argument. >>>>> >>>>> This program was working very well until I upgraded FreeSwitch. Now >>>>> the program seems to hang when it tries to write out the first converted >>>>> files. I have to KILL the program from another terminal. It leaves 2 0 >>>>> byte files with .PCMU and .G729 extension. >>>>> >>>>> I have run through a GIT bisect to find where things broke. The last >>>>> working version is Version 1.6.16 git ae1cdce 2017-04-11. >>>>> The first broken version is 38621e47bad3b63f03a0a27f6ca9ed92f6969032. >>>>> >>>>> I then used GIT DIFF to see if I could see what had happened. There >>>>> are very few modifications between these two commits: >>>>> >>>>> root at fs3:/usr/src/freeswitch.git# git diff ae1cdce >>>>> 38621e47bad3b63f03a0a27f6ca9ed92f6969032 >>>>> diff --git a/src/include/switch_module_interfaces.h >>>>> b/src/include/switch_module_interfaces.h >>>>> index e0a5c20..7ca027d 100644 >>>>> --- a/src/include/switch_module_interfaces.h >>>>> +++ b/src/include/switch_module_interfaces.h >>>>> @@ -329,6 +329,11 @@ typedef struct switch_mm_s { >>>>> switch_video_profile_t vprofile; >>>>> switch_video_encode_speed_t vencspd; >>>>> uint8_t try_hardware_encoder; >>>>> + int scale_w; >>>>> + int scale_h; >>>>> + switch_img_fmt_t fmt; >>>>> + char *auth_username; >>>>> + char *auth_password; >>>>> } switch_mm_t; >>>>> >>>>> /*! an abstract representation of a file handle (some parameters >>>>> based on compat with libsndfile) */ >>>>> diff --git a/src/mod/applications/mod_av/avformat.c >>>>> b/src/mod/applications/mod_av/avformat.c >>>>> index b944625..4b92801 100644 >>>>> --- a/src/mod/applications/mod_av/avformat.c >>>>> +++ b/src/mod/applications/mod_av/avformat.c >>>>> @@ -906,7 +906,7 @@ SWITCH_STANDARD_APP(record_av_function) >>>>> char codec_str[256]; >>>>> const AVCodecDescriptor *desc; >>>>> >>>>> - if (!strncmp(data, "rtmp://", 7)) { >>>>> + if (!strncmp(data, "rtmp://", 7) || !strncmp(data, >>>>> "rtsp://", 7)) { >>>>> fmt->video_codec = AV_CODEC_ID_H264; >>>>> fmt->audio_codec = AV_CODEC_ID_AAC; >>>>> } >>>>> @@ -1694,9 +1694,20 @@ static switch_status_t >>>>> av_file_open(switch_file_handle_t *handle, const char *pa >>>>> return SWITCH_STATUS_GENERR; >>>>> } else if (handle->stream_name && >>>>> (!strcasecmp(handle->stream_name, "rtmp") || >>>>> !strcasecmp(handle->stream_name, "youtube"))) { >>>>> format = "flv"; >>>>> - switch_snprintf(file, sizeof(file), "rtmp://%s", >>>>> path); >>>>> + >>>>> + // meh really silly format for the user / pass >>>>> libav..... >>>>> + if (handle->mm.auth_username && >>>>> handle->mm.auth_password) { >>>>> + switch_snprintf(file, sizeof(file), "rtmp://%s >>>>> pubUser=%s pubPasswd=%s flashver=FMLE/3.0", path, handle->mm.auth_username, >>>>> handle->mm.auth_password); >>>>> + } else { >>>>> + switch_snprintf(file, sizeof(file), "rtmp://%s", >>>>> path); >>>>> + } >>>>> + >>>>> + } else if (handle->stream_name && >>>>> !strcasecmp(handle->stream_name, "rtsp")) { >>>>> + format = "rtsp"; >>>>> + switch_snprintf(file, sizeof(file), "rtsp://%s", >>>>> path); >>>>> } >>>>> >>>>> + >>>>> ext++; >>>>> >>>>> if ((context = (av_file_context_t >>>>> *)switch_core_alloc(handle->memory_pool, sizeof(av_file_context_t))) == 0) { >>>>> @@ -1783,7 +1794,7 @@ static switch_status_t >>>>> av_file_open(switch_file_handle_t *handle, const char *pa >>>>> if (fmt->video_codec != AV_CODEC_ID_NONE) { >>>>> const AVCodecDescriptor *desc; >>>>> >>>>> - if ((handle->stream_name && >>>>> (!strcasecmp(handle->stream_name, "rtmp") || >>>>> !strcasecmp(handle->stream_name, "youtube")))) { >>>>> + if ((handle->stream_name && >>>>> (!strcasecmp(handle->stream_name, "rtmp") || >>>>> !strcasecmp(handle->stream_name, "rtsp") || >>>>> !strcasecmp(handle->stream_name, "youtube")))) { >>>>> >>>>> if (fmt->video_codec != AV_CODEC_ID_H264 ) { >>>>> fmt->video_codec = AV_CODEC_ID_H264; >>>>> // force H264 >>>>> @@ -2525,6 +2536,7 @@ SWITCH_MODULE_LOAD_FUNCTION(mod_avformat_load) >>>>> >>>>> supported_formats[i++] = "av"; >>>>> supported_formats[i++] = "rtmp"; >>>>> + supported_formats[i++] = "rtsp"; >>>>> supported_formats[i++] = "mp4"; >>>>> supported_formats[i++] = "m4a"; >>>>> supported_formats[i++] = "mov"; >>>>> diff --git a/src/mod/applications/mod_av/mod_av.c >>>>> b/src/mod/applications/mod_av/mod_av.c >>>>> index 3d3bd82..141fcdc 100644 >>>>> --- a/src/mod/applications/mod_av/mod_av.c >>>>> +++ b/src/mod/applications/mod_av/mod_av.c >>>>> @@ -93,7 +93,7 @@ int mod_av_lockmgr_cb(void **m, enum AVLockOp op) >>>>> static void log_callback(void *ptr, int level, const char *fmt, >>>>> va_list vl) >>>>> { >>>>> switch_log_level_t switch_level = SWITCH_LOG_DEBUG; >>>>> - >>>>> + return; >>>>> /* naggy messages */ >>>>> if (level == AV_LOG_DEBUG || level == AV_LOG_WARNING) return; >>>>> >>>>> diff --git a/src/switch_core_file.c b/src/switch_core_file.c >>>>> index 46ee539..aff9442 100644 >>>>> --- a/src/switch_core_file.c >>>>> +++ b/src/switch_core_file.c >>>>> @@ -177,6 +177,14 @@ SWITCH_DECLARE(switch_status_t) >>>>> switch_core_perform_file_open(const char *file, >>>>> fh->mm.try_hardware_encoder = switch_true(val); >>>>> } >>>>> >>>>> + if ((val = switch_event_get_header(fh->params, >>>>> "auth_username"))) { >>>>> + fh->mm.auth_username = >>>>> switch_core_strdup(fh->memory_pool, val); >>>>> + } >>>>> + >>>>> + if ((val = switch_event_get_header(fh->params, >>>>> "auth_password"))) { >>>>> + fh->mm.auth_password = >>>>> switch_core_strdup(fh->memory_pool, val); >>>>> + } >>>>> + >>>>> if ((val = switch_event_get_header(fh->params, >>>>> "fps"))) { >>>>> float ftmp = atof(val); >>>>> if (ftmp > 0.0f) { >>>>> (END) >>>>> >>>>> >>>>> I believe that it is the changes in switch_core_file.c that break my >>>>> utility program, but I can't be sure. Just o add some detail, I have a >>>>> Digium TCE400 card in my system which handles the G729 conversion. >>>>> >>>>> I have no idea of what those auth_username and auth_password >>>>> parameters are, but that seems to break things for me. Any idea of what I >>>>> can do to get things working again? >>>>> >>>>> I have created a PasteBin with the code of my fs_encode derived >>>>> utility: https://pastebin.freeswitch.org/view/a3d64c69 It is pretty >>>>> short. >>>>> >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at 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 at 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 ♬ @anthmfs ♬ @FreeSWITCH ♬ >>>> >>>> ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ >>>> http://twitter.com/FreeSWITCH >>>> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ >>>> * >>>> >>>> ClueCon Weekly Development Call >>>> ☎ sip:888 at conference.freeswitch.org ☎ +19193869900 <(919)%20386-9900> >>>> >>>> https://www.youtube.com/watch?v=9XXgW34t40s >>>> https://www.youtube.com/watch?v=NLaDpGQuZDA >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at 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 at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> >>> -- >>> Guillermo Ruiz Camauer >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at 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 at 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 ♬ @anthmfs ♬ @FreeSWITCH ♬ >> >> ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ >> http://twitter.com/FreeSWITCH >> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ >> * >> >> ClueCon Weekly Development Call >> ☎ sip:888 at conference.freeswitch.org ☎ +19193869900 >> >> https://www.youtube.com/watch?v=9XXgW34t40s >> https://www.youtube.com/watch?v=NLaDpGQuZDA >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Guillermo Ruiz Camauer > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at 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 ♬ @anthmfs ♬ @FreeSWITCH ♬ ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ * ClueCon Weekly Development Call ☎ sip:888 at conference.freeswitch.org ☎ +19193869900 https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA -------------- next part -------------- An HTML attachment was scrubbed... URL: From jaco at grandcom.co.za Fri Jun 23 21:19:04 2017 From: jaco at grandcom.co.za (Jaco Jacobs) Date: Fri, 23 Jun 2017 21:19:04 +0000 Subject: [Freeswitch-users] att_xfer or other method for mod_callcenter queue to queue transfer Message-ID: I've been using mod_callcenter in what I can imagine is not the originally intended use case. It's worked so far but now I've run into a problem and being new to FS I've gotten stuck. I originate a call to an extension on another voice system (non FS), and then use Mod_callcenter uuid_standby to keep the agent ready in the queue. I then make an outbound call and put the call into the queue ( like a dialer or click to dial type setup). This all works great, the problem comes when I want to transfer. I've tried uuid-transfer, tried to park the call etc., as soon as I try this the outbound call drops. The closest I've gotten is to use att_xfer directly to an extension on the other non FS voice system, this works fine. The problem is if I want to transfer to another queue, which is a function I definitely want. Any assistance would be greatly appreciated. -------------- next part -------------- An HTML attachment was scrubbed... URL: From admin at blindi.net Sat Jun 24 15:55:03 2017 From: admin at blindi.net (Thomas Hoellriegel) Date: Sat, 24 Jun 2017 17:55:03 +0200 (CEST) Subject: [Freeswitch-users] Question: Skype 4.3 support endet! alternate to Fs? Message-ID: Hi all, Microsoft remove all older skypeclients on 07.01.2017. The problem: skype is not working after this date. mod_skypopen support only skype 4.3.37. I don't found a manual to use fs on latest skype versions. Do your can help please? Do your have a way after on Jul 2017? thanks. --------------- Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: http://www.blindi.net/callback homepage: http://www.blindi.net blinde-misc mailingliste für blinde. anmeldung unter: http://www.blindi.net/mailman/listinfo/blinde-misc From gmaruzz at gmail.com Sat Jun 24 16:22:13 2017 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Sat, 24 Jun 2017 18:22:13 +0200 Subject: [Freeswitch-users] Question: Skype 4.3 support endet! alternate to Fs? In-Reply-To: References: Message-ID: On 24 June 2017 at 17:55, Thomas Hoellriegel wrote: > Hi all, > > Microsoft remove all older skypeclients on 07.01.2017. > The problem: skype is not working after this date. > mod_skypopen support only skype 4.3.37. > I don't found a manual to use fs on latest skype versions. > > Do your can help please? > Do your have a way after on Jul 2017? > NOT. http://lists.freeswitch.org/pipermail/freeswitch-users/2017-June/126571.html -------------- next part -------------- An HTML attachment was scrubbed... URL: From jaco at grandcom.co.za Sat Jun 24 19:09:25 2017 From: jaco at grandcom.co.za (Jaco Jacobs) Date: Sat, 24 Jun 2017 19:09:25 +0000 Subject: [Freeswitch-users] att_xfer or other method for mod_callcenter queue to queue transfer In-Reply-To: References: Message-ID: <71afb4932271407ca46362a1b48b5a23@grandcom.co.za> If I'm being stupid I apologise, Is there a way I can originate a dummy call to a queue, then use attxfer to that call. This is what I did to transfer the call directly to the extension on the other system. uuid_broadcast 57176bf7-111a-40c6-b1c6-9afcf11111a att_xfer::sofia/gateway/toEXTPBX/8005 I have this in a dialplan to connect a call I originate via API to a queue (I know there is an unnecessary step here but I use to do some checks for unanswered calls and this is a remnent design I'll clean up later): #I created a unique queue per agent for now, which I'm using for click to dial as mod_callcenter solves a lot of things I need for the wallboard I don't know the right terminology here, my background is in Cisco Callobarotion, this is something new I'm trying so excuse me if I'm using the wrong terminology etc. But is there a way I can call the above dialplan via API with a fake call and I will then have something to transfer to. Or is there something I can change this string to that will alow me to transfer to the queue : uuid_broadcast 57176bf7-111a-40c6-b1c6-9afcf1111a att_xfer::sofia/gateway/toEXTPBX/8005 From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jaco Jacobs Sent: Friday, 23 June 2017 11:19 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] att_xfer or other method for mod_callcenter queue to queue transfer I've been using mod_callcenter in what I can imagine is not the originally intended use case. It's worked so far but now I've run into a problem and being new to FS I've gotten stuck. I originate a call to an extension on another voice system (non FS), and then use Mod_callcenter uuid_standby to keep the agent ready in the queue. I then make an outbound call and put the call into the queue ( like a dialer or click to dial type setup). This all works great, the problem comes when I want to transfer. I've tried uuid-transfer, tried to park the call etc., as soon as I try this the outbound call drops. The closest I've gotten is to use att_xfer directly to an extension on the other non FS voice system, this works fine. The problem is if I want to transfer to another queue, which is a function I definitely want. Any assistance would be greatly appreciated. -------------- next part -------------- An HTML attachment was scrubbed... URL: From infos at madovsky.org Sun Jun 25 09:52:25 2017 From: infos at madovsky.org (Madovsky) Date: Sun, 25 Jun 2017 02:52:25 -0700 Subject: [Freeswitch-users] postgresql ssl Message-ID: <96a21c83-ef2d-5f59-4260-58685c3e8c1d@madovsky.org> Hi folks, is there any special syntax to connect postgresql in ssl mode from pgsql://? thanks From krice at freeswitch.org Sun Jun 25 13:51:43 2017 From: krice at freeswitch.org (Ken Rice) Date: Sun, 25 Jun 2017 08:51:43 -0500 Subject: [Freeswitch-users] postgresql ssl In-Reply-To: <96a21c83-ef2d-5f59-4260-58685c3e8c1d@madovsky.org> References: <96a21c83-ef2d-5f59-4260-58685c3e8c1d@madovsky.org> Message-ID: No special syntax required... your pgsql server must have ssl support enabled and your libpq install must have been built with ssl support enabled. From there it is automattic -----Original Message----- From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Madovsky Sent: Sunday, June 25, 2017 4:52 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] postgresql ssl Hi folks, is there any special syntax to connect postgresql in ssl mode from pgsql://? thanks _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From bipin at xbipin.com Sun Jun 25 15:10:04 2017 From: bipin at xbipin.com (Bipin Patel) Date: Sun, 25 Jun 2017 19:10:04 +0400 Subject: [Freeswitch-users] keep limit counter active across dialplans when used in attended transfers Message-ID: hi, i have a requirement where i need to generate a different ringtone if that ext is on another call and i do it using limit by setting a counter for the caller and callee and then checking the counter if it has a value of 1 which would mean that ext is on anohter call so generate a different ringtone and this work fine. The problem being if A calls B and B attended transfers the call to C then the counter for C is incremented in the features dialplan but it doesnt reflect in the internal dialplan and i read that active limits dont remain global across dialplans so is there anyway to keep the limit counter active. i tried limit_ignore_transfer but that doesnt seem to work, im on the FS master Below is my internal and features dialplan, any pointers would be helpful INTERNAL DIALPLAN FEATURES DIALPLAN -- Regards, Bipin ------------------------------------------------------------------------ -------------- next part -------------- An HTML attachment was scrubbed... URL: From magnus.kelly at gmail.com Sun Jun 25 16:14:49 2017 From: magnus.kelly at gmail.com (Magnus Kelly) Date: Sun, 25 Jun 2017 17:14:49 +0100 Subject: [Freeswitch-users] Refer with replaces - how to configure to support Message-ID: Hello all, I am trying to configure the FS dial plan to fully support "Refer with Replaces" ? Call case is FS establishes the call successfully with an upstream server, but when the user behind the upstream server performs a transfer, a Refer with replaces is generated and sent to FS, FS accepts the request, but post the 202 the expected invite (FS to upstream) in support of replaces does not occur and then the transfer fails. My assumption is I am missing the correct configuration for Refer with Replaces, I have spent some time searching web but not yet found the correct way to handle this. My assumption is FS fully supports "Refer with Replaces" but confirmation and any tips on configuration most welcome. Appreciate any insight. Thanks Magnus -------------- next part -------------- An HTML attachment was scrubbed... URL: From vma at vallimamod.org Sun Jun 25 16:41:14 2017 From: vma at vallimamod.org (Vallimamod Abdullah) Date: Sun, 25 Jun 2017 18:41:14 +0200 Subject: [Freeswitch-users] keep limit counter active across dialplans when used in attended transfers In-Reply-To: References: Message-ID: Hi, Is there any reason why you don't use the hash application directly instead of limit? IMHO it is more suited to your use case. https://freeswitch.org/confluence/display/FREESWITCH/mod_hash Best Regards, -- Vallimamod Abdullah SIP Solutions vma at sipsolutions.fr . > On 25 Jun 2017, at 17:10, Bipin Patel wrote: > > hi, > > i have a requirement where i need to generate a different ringtone if that ext is on another call and i do it using limit by setting a counter for the caller and callee and then checking the counter if it has a value of 1 which would mean that ext is on anohter call so generate a different ringtone and this work fine. > > The problem being if A calls B and B attended transfers the call to C then the counter for C is incremented in the features dialplan but it doesnt reflect in the internal dialplan and i read that active limits dont remain global across dialplans so is there anyway to keep the limit counter active. i tried limit_ignore_transfer but that doesnt seem to work, im on the FS master > > Below is my internal and features dialplan, any pointers would be helpful > > > INTERNAL DIALPLAN > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > FEATURES DIALPLAN > > > > > > > > > > > > > > > > -- > Regards, > Bipin > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From xxxman2008 at 126.com Sun Jun 25 18:28:59 2017 From: xxxman2008 at 126.com (Raymond) Date: Mon, 26 Jun 2017 02:28:59 +0800 (CST) Subject: [Freeswitch-users] Javascript FileIO In-Reply-To: References: <2e9e2174.97ca.15cd41840cd.Coremail.xxxman2008@126.com> Message-ID: <7fca2474.3e0.15ce0839672.Coremail.xxxman2008@126.com> OK， firstly ，where's the new documentation of new V8 js module ? Secondly, I think, loop read small piece of file will cause too much I/O request . SO , i usually do read in following way. open the file ----> get file size --> read the whole file ( OS will charge the "one time " read size). and in real-life , real-project , there's not so many big files. So, just open the file ,and read all. in 99.99% situation , it's safe. in some situation , you cann't open the "incomplete file" , you even cann't "see" them. There's too much detail when we talk about file read ,we can discuss file-system in rest of our life. For Faruq , i think just read whole file is ok . if you warry about "incomplete file" ,just wait 1 second ,and read the file size again. it's may casuse delay ,but useful if you have lots of files to read . In my opinion, when linux do file read ,it read 1 block one time . A block is a sequence of bit or Bytes with a fixed length ie 512 bytes, 4kB, 8kB, 16kB, 32kB etc. So , please not use strange number as your read size. it's not a good idea. And ......I really hate "while (.....)" code ,it's really really bad. Raymond At 2017-06-24 04:00:36, "Anthony Minessale" wrote: That wiki documentation is from the old spidermonkey js module not the new v8 one. I believe that putting a very large number will not cause any problems as the data read cannot exceed the total bytes in the file so it should not be a problem. You can always do an append loop. Its usually a good practice to have a precise max size to read because what if the file was accidentally 2gb. On Fri, Jun 23, 2017 at 3:36 AM, Raymond wrote: Hi , This will help you . https://wiki.freeswitch.org/wiki/File Raymond At 2017-06-22 21:49:33, "Faruq Ahmad" wrote: Hi, I have a FS dialplan that fetches callflow from a JSON file. File is read using var JsonFd = new FileIO(JsonPath, 'r'); JsonFd.read(2048); result = JsonFd.data(); var menu = JSON.parse(result), Is there anyway I can read the whole file in one attempt, i.e. when a smaller size was given to the .read() function it wouldn't read the complete file. I have increased the value of size for read function however my concern is in the long run file size might increase and parse would get an incomplete JSON. Is there anyway I can get filesize from FileIO object or detect EOF from the read buffer to make sure the whole file is loaded? Also if I increase the size for read fucntion buffer way over the estimated filesizes, is there any guarantee that no garbage values will be read from the disk after the EOF? _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at 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 at 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 ♬ @anthmfs ♬ @FreeSWITCH ♬ ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ http://freeswitch.org/g+ ClueCon Weekly Development Call ☎ sip:888 at conference.freeswitch.org ☎ +19193869900 https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA -------------- next part -------------- An HTML attachment was scrubbed... URL: From anthony.minessale at gmail.com Sun Jun 25 19:17:00 2017 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 25 Jun 2017 19:17:00 +0000 Subject: [Freeswitch-users] Javascript FileIO In-Reply-To: <7fca2474.3e0.15ce0839672.Coremail.xxxman2008@126.com> References: <2e9e2174.97ca.15cd41840cd.Coremail.xxxman2008@126.com> <7fca2474.3e0.15ce0839672.Coremail.xxxman2008@126.com> Message-ID: On Sun, Jun 25, 2017 at 1:29 PM Raymond wrote: > > OK， firstly ，where's the new documentation of new V8 js module ? > Use the source luke. Maybe you can join the volunteer docs team and help them. The module itself was donated by a community member. > Secondly, I think, loop read small piece of file will cause too much > I/O request . SO , i usually do read in following way. > open the file ----> get file size --> read the whole file ( OS will charge > the "one time " read size). > That is completely relevant to the size of the file and the size of the chunk. There is no one answer. > and in real-life , real-project , there's not so many big files. So, just > open the file ,and read all. in 99.99% situation , it's safe. > Um, ok..... That does not sound very wise but sure. I'd say At least test the size of the file first. > in some situation , you cann't open the "incomplete file" , you even > cann't "see" them. There's too much detail when we talk about file read > ,we can discuss file-system in rest of our life. > For Faruq , i think just read whole file is ok . if you warry about "incomplete > file" ,just wait 1 second ,and read the file size again. it's may casuse > delay ,but useful if you have lots of files to read . > > In my opinion, when linux do file read ,it read 1 block one time . A > block is a sequence of bit or Bytes with a fixed length ie 512 bytes, 4kB, > 8kB, 16kB, 32kB etc. So , please not use strange number as your read size. > it's not a good idea. And ......I really hate "while (.....)" code ,it's > really really bad. > Shaming coding constructs, in my opinion, is the path to mistakes. Every code element exists for a reason and there is always a case for any of them. I hate coding construct x is not helpful. > Raymond > > > At 2017-06-24 04:00:36, "Anthony Minessale" > wrote: > > That wiki documentation is from the old spidermonkey js module not the new > v8 one. > > I believe that putting a very large number will not cause any problems as > the data read cannot exceed the total bytes in the file so it should not be > a problem. > You can always do an append loop. Its usually a good practice to have a > precise max size to read because what if the file was accidentally 2gb. > > On Fri, Jun 23, 2017 at 3:36 AM, Raymond wrote: > >> Hi , >> >> This will help you . https://wiki.freeswitch.org/wiki/File >> >> Raymond >> >> At 2017-06-22 21:49:33, "Faruq Ahmad" wrote: >> >> Hi, >> >> I have a FS dialplan that fetches callflow from a JSON file. File is read >> using >> >> var JsonFd = new FileIO(JsonPath, 'r'); >> JsonFd.read(2048); >> result = JsonFd.data(); >> var menu = JSON.parse(result), >> >> Is there anyway I can read the whole file in one attempt, i.e. when a >> smaller size was given to the .read() function it wouldn't read the >> complete file. I have increased the value of size for read function however >> my concern is in the long run file size might increase and parse would get >> an incomplete JSON. >> >> Is there anyway I can get filesize from FileIO object or detect EOF from >> the read buffer to make sure the whole file is loaded? >> Also if I increase the size for read fucntion buffer way over the >> estimated filesizes, is there any guarantee that no garbage values will be >> read from the disk after the EOF? >> >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at 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 ♬ @anthmfs ♬ @FreeSWITCH ♬ > > ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ > http://twitter.com/FreeSWITCH > ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ > * > > ClueCon Weekly Development Call > ☎ sip:888 at conference.freeswitch.org ☎ +19193869900 > > https://www.youtube.com/watch?v=9XXgW34t40s > https://www.youtube.com/watch?v=NLaDpGQuZDA > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at 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 ♬ @anthmfs ♬ @FreeSWITCH ♬ ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ * ClueCon Weekly Development Call ☎ sip:888 at conference.freeswitch.org ☎ +19193869900 https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA -------------- next part -------------- An HTML attachment was scrubbed... URL: From bipin at xbipin.com Sun Jun 25 19:54:16 2017 From: bipin at xbipin.com (Bipin Patel) Date: Sun, 25 Jun 2017 23:54:16 +0400 Subject: [Freeswitch-users] keep limit counter active across dialplans when used in attended transfers In-Reply-To: References: Message-ID: <15ce0d1aa40.279b.b07ebdf329620b8089087c7205b03f01@xbipin.com> Hi, Well I could use it but with limit the counter decrementing etc is automatic so it's much simpler and reduces the dialplan required. In the dialplan I provided how do you think I could use it directly without adding any more extensions in the two dialplan, any help would be appreciated. What I need to do is increment counter for caller and callee as well as callee transferred to and check that counter before bridging so change ring back can be called. On June 25, 2017 8:43:29 PM Vallimamod Abdullah wrote: > Hi, > > Is there any reason why you don't use the hash application directly instead > of limit? > IMHO it is more suited to your use case. > > https://freeswitch.org/confluence/display/FREESWITCH/mod_hash > > Best Regards, > -- > Vallimamod Abdullah > SIP Solutions > vma at sipsolutions.fr > . > > >> On 25 Jun 2017, at 17:10, Bipin Patel wrote: >> >> hi, >> >> i have a requirement where i need to generate a different ringtone if that >> ext is on another call and i do it using limit by setting a counter for the >> caller and callee and then checking the counter if it has a value of 1 >> which would mean that ext is on anohter call so generate a different >> ringtone and this work fine. >> >> The problem being if A calls B and B attended transfers the call to C then >> the counter for C is incremented in the features dialplan but it doesnt >> reflect in the internal dialplan and i read that active limits dont remain >> global across dialplans so is there anyway to keep the limit counter >> active. i tried limit_ignore_transfer but that doesnt seem to work, im on >> the FS master >> >> Below is my internal and features dialplan, any pointers would be helpful >> >> >> INTERNAL DIALPLAN >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> > data="transfer_ringback=%(250,250,400);%(250,3500,400)"/> >> >> >> >> >> >> >> >> >> >> > data="transfer_ringback=%(400,200,400,375);%(400,2000,400,375)"/> >> >> > data="ringback=%(400,200,400,375);%(400,2000,400,375)"/> >> >> >> >> >> >> >> >> >> FEATURES DIALPLAN >> >> >> >> > data="transfer_ringback=%(400,200,400,375);%(400,2000,400,375)"/> >> >> > data="ringback=%(400,200,400,375);%(400,2000,400,375)"/> >> >> >> >> >> >> >> >> >> >> -- >> Regards, >> Bipin >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From magnus.kelly at gmail.com Sun Jun 25 21:10:55 2017 From: magnus.kelly at gmail.com (Magnus Kelly) Date: Sun, 25 Jun 2017 22:10:55 +0100 Subject: [Freeswitch-users] Refer with replaces - how to configure to support Message-ID: Hello All, Further to prior question on support for Refer with replaces configuration question - the ladder diagram below is the desired sequence - currently fails at line 45 This is one of the call cases required to interop with MS Skype4Biz of which I am curious if this can be handled as it seems the Refer C target is not revealed until after the "parking" Invite is sent Example Refer is below ------------------------------------------------------------------------ REFER sip:gw+s4btrunk at 10.113.3.4:5066;transport=udp;gw=s4btrunk SIP/2.0 FROM: ;epid=ED60F08DC5;tag=6acfb33460 TO: ;tag=r3cp3tF36UmZg CSEQ: 2 REFER CALL-ID: 78dff094-ca55-1235-8c91-000c29a74006 MAX-FORWARDS: 70 VIA: SIP/2.0/TCP 10.113.8.32:5068;branch=z9hG4bK433d335f CONTACT: CONTENT-LENGTH: 0 REFER-TO: USER-AGENT: RTCC/6.0.0.0 MediationServer ------------------------------------------------------------------------ 1PSTN 2 S4B/Mediation Server PSTN/FS1 -IN 2 | | | 3 | | (1) INVITE | 4 | |<----------------------------------------------| 5 | | | 6 | | (2) 100 Trying | 7 | |---------------------------------------------->| 8 | | | 9 | | (3) 183 Session Progress (Dialog 1) | 10 | |---------------------------------------------->| 11 | | | 12 | | (4) PRACK | 13 | |<----------------------------------------------| 14 | | | 15 | | (5) 200 OK (PRACK) | 16 | |---------------------------------------------->| 17 | | | 18 | | Early Media | 19 | |==============================================>| 20 | | | 21 | | (6) 200 OK (Dialog 1) | 22 | |---------------------------------------------->| 23 | | | 24 | | (7) ACK | 25 | |<----------------------------------------------| 26 | | | 27 | | Bidirectional RTP | 28 | |<=============================================>| 29 | | | 30 | | (8) INVITE(Hold) | 31 | |---------------------------------------------->| 32 | | | 33 | | (9) 200 OK (hold) | 34 | |<----------------------------------------------| 35 | | | 36 | | (10) ACK | 37 | |---------------------------------------------->| 38 | | | 39 | | (11) REFER | 40 | |---------------------------------------------->| 41 | | | 42 | | (12) 202 Accepted (OK till here) | 43 | |<----------------------------------------------| 44 | | | 45 | | (13) INVITE(Replaces, Dialog 2) (Not sent)| 46 | |<----------------------------------------------| 47 | | | 48 | | (14) 183 Session Progress(Dialog 2) | 49 | |---------------------------------------------->| 50 | | | 51 | | (15) NOTIFY (Trying) | 52 | |<----------------------------------------------| 53 | | | 54 | | (16) 200 OK (NOTIFY) | 55 | |---------------------------------------------->| 56 | | | 57 | (17) INVITE(Dialog 3) | | 58 |<---------------------------------| | 59 | | | 60 | (18) 183 Session Progress | | 61 |--------------------------------->| | 62 | | | 63 | | (19) 183 Session Progress(Dialog 2) | 64 | |---------------------------------------------->| 65 | | | 66 | | (20) NOTIFY (Trying) | 67 | |<----------------------------------------------| 68 | | | 69 | | (21) 200 OK (NOTIFY) | 70 | |---------------------------------------------->| 71 | | | 72 | (22) 200 OK(Dialog 3) | | 73 |--------------------------------->| | 74 | | | 75 | | (23) 200 OK(Dialog 2) | 76 | |---------------------------------------------->| 77 | | | 78 | | (24) NOTIFY | 79 | |<----------------------------------------------| 80 | | | 81 | | (25) 200 OK (NOTIFY) | 82 | |---------------------------------------------->| 83 | | | 84 | | (26) BYE(Dialog 1) | 85 | |<----------------------------------------------| 86 | | | 87 | | (27) 200 OK (BYE, Dialog 1) | 88 | |---------------------------------------------->| On 25 June 2017 at 17:14, Magnus Kelly wrote: > Hello all, > > I am trying to configure the FS dial plan to fully support "Refer with > Replaces" ? > > Call case is FS establishes the call successfully with an upstream server, > but when the user behind the upstream server performs a transfer, a Refer > with replaces is generated and sent to FS, FS accepts the request, but post > the 202 the expected invite (FS to upstream) in support of replaces does > not occur and then the transfer fails. > > My assumption is I am missing the correct configuration for Refer with > Replaces, I have spent some time searching web but not yet found the > correct way to handle this. My assumption is FS fully supports "Refer with > Replaces" but confirmation and any tips on configuration most welcome. > > Appreciate any insight. > > Thanks > Magnus > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: From arsen.semionov at gmail.com Mon Jun 26 10:35:15 2017 From: arsen.semionov at gmail.com (Arsen) Date: Mon, 26 Jun 2017 13:35:15 +0300 Subject: [Freeswitch-users] SKYPE: mod_skypopen END-OF-LIFE In-Reply-To: References: Message-ID: Really sad to hear this. It was a great module, we have been able to create couple of nice services using it. Thank you Giovanni for all work and support. Cheers, On Fri, Jun 23, 2017 at 4:15 PM, Giovanni Maruzzelli wrote: > > On 23 June 2017 at 15:07, Volodymyr Fedorov wrote: > >> Thanks for your work Giovanni, >> But I have a question. Did somebody look to ability to bridge webrtc >> Skype client and Freeswitch ? >> Current version Skype for linux is "just" chromium page is`t it. For chat >> protocol exist implementation for Web-skype: >> https://github.com/EionRobb/skype4pidgin/tree/master/skypeweb >> >> > if/when there will be an endorsed, legal, free and redistributable way to > do it, I will be happy to look into that and try. > > Definitely not if its legal part is less than clear, and if is not > endorsed by Microsoft (eg, not an hack that will stop to work when they > change something). > > That said, good luck to others > > > > > > >> Thanks. >> >> >> On Fri, Jun 23, 2017 at 11:29 AM, Giovanni Maruzzelli >> wrote: >> >>> Ciao Mitul !!!! >>> >>> A friendly and dear hug ! >>> -giovanni >>> >>> >>> >>> On 23 June 2017 at 11:26, Mitul Limbani wrote: >>> >>>> Sad to see this die :( >>>> >>>> Mitul >>>> >>>> Regards, >>>> Mitul Limbani, >>>> Business Head, >>>> Enterux Solutions Pvt. Ltd. >>>> 110 Reena Complex, Opp. Nathani Steel, >>>> Vidyavihar (W), Mumbai - 400 086. India >>>> http://www.enterux.com/ >>>> http://www.entvoice.com/ >>>> email: mitul at enterux.in >>>> DID: +91-22-71967196 <+91%2022%207196%207196> >>>> Cell: +91-9820332422 <+91%2098203%2032422> >>>> >>>> On Fri, Jun 23, 2017 at 2:16 PM, Giovanni Maruzzelli >>> > wrote: >>>> >>>>> Hello FreeSWITCHers, >>>>> >>>>> >>>>> Microsoft decided to kill all old Skype clients July the 1st, and new >>>>> clients do not support APIs to interact with calls. >>>>> >>>>> So, no way mod_skypopen can continue to work. >>>>> >>>>> Has been a nice trip, a lot has changed in the Real Time Communication >>>>> world since first chan_skypiax, then mod_skypopen bridged Skype and open >>>>> standard calls. >>>>> >>>>> mod_skypopen sez: "GOODBYE WORLD!" >>>>> >>>>> >>>>> -giovanni >>>>> >>>>> >>>>> >>>>> -- >>>>> >>>>> Sincerely, >>>>> >>>>> Giovanni Maruzzelli >>>>> OpenTelecom.IT >>>>> cell: +39 347 266 56 18 >>>>> >>>>> ____________________________________________________________ >>>>> _____________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at 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 at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>>> switch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> ____________________________________________________________ >>>> _____________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at 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 at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>> switch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> >>> -- >>> >>> Sincerely, >>> >>> Giovanni Maruzzelli >>> OpenTelecom.IT >>> cell: +39 347 266 56 18 >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at 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 at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Best regards, >> Volodymyr >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at 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 > OpenTelecom.IT > cell: +39 347 266 56 18 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Regards, Arsen Semionov Eurolan.info cell: +442035198881 -------------- next part -------------- An HTML attachment was scrubbed... URL: From kkothari157 at gmail.com Mon Jun 26 08:32:41 2017 From: kkothari157 at gmail.com (Ketan Kothari) Date: Mon, 26 Jun 2017 14:02:41 +0530 Subject: [Freeswitch-users] Verto configure with domain name Message-ID: Hello Guys, I setup verto communicator on our server but now we want to run it on our domain name also can anyone help to remove its default port 9001? Please find below image of logs of verto http://prntscr.com/fo7o1c Please help me on this. -------------- next part -------------- An HTML attachment was scrubbed... URL: From italo at freeswitch.org Mon Jun 26 10:47:01 2017 From: italo at freeswitch.org (=?UTF-8?B?w410YWxvIFJvc3Np?=) Date: Mon, 26 Jun 2017 10:47:01 +0000 Subject: [Freeswitch-users] Verto configure with domain name In-Reply-To: References: Message-ID: You need to build verto communicator (grunt build). This will give you a folder called dist, just copy the folder to a directory where your web server can see and configure your web server as you want. This is described in verto communicator Confluence page. Em seg, 26 de jun de 2017 às 07:42, Ketan Kothari escreveu: > Hello Guys, > > I setup verto communicator on our server but now we want to run it on our > domain name also can anyone help to remove its default port 9001? > > Please find below image of logs of verto > http://prntscr.com/fo7o1c > > Please help me on this. > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From gmaruzz at gmail.com Mon Jun 26 11:08:57 2017 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Mon, 26 Jun 2017 13:08:57 +0200 Subject: [Freeswitch-users] SKYPE: mod_skypopen END-OF-LIFE In-Reply-To: References: Message-ID: On 26 June 2017 at 12:35, Arsen wrote: > Really sad to hear this. It was a great module, we have been able to > create couple of nice services using it. > Thank you for your nice words, Arsen, very appreciated! -giovanni -------------- next part -------------- An HTML attachment was scrubbed... URL: From bipin at xbipin.com Mon Jun 26 13:59:03 2017 From: bipin at xbipin.com (Bipin Patel) Date: Mon, 26 Jun 2017 17:59:03 +0400 Subject: [Freeswitch-users] keep limit counter active across dialplans when used in attended transfers In-Reply-To: <15ce0d1aa40.279b.b07ebdf329620b8089087c7205b03f01@xbipin.com> References: <15ce0d1aa40.279b.b07ebdf329620b8089087c7205b03f01@xbipin.com> Message-ID: <752b7178-f635-a544-05a6-df985bfa851f@xbipin.com> hi, i was testing mod_hash but how do we delete the key when the caller has hung up causing originator cancel because the dialplan doesnt execute any further, it does when the call is answered then hungup. Im assuming mod_hash works across dialplans because i didnt find anything in docs that would suggest it wont Regards, Bipin ------------------------------------------------------------------------ -------- Original Message -------- Subject: Re: [Freeswitch-users] keep limit counter active across dialplans when used in attended transfers From: Bipin Patel To: FreeSWITCH Users Help Date: 6/25/2017, 11:54:16 PM > > Hi, > > Well I could use it but with limit the counter decrementing etc is > automatic so it's much simpler and reduces the dialplan required. > > In the dialplan I provided how do you think I could use it directly > without adding any more extensions in the two dialplan, any help would > be appreciated. What I need to do is increment counter for caller and > callee as well as callee transferred to and check that counter before > bridging so change ring back can be called. > > On June 25, 2017 8:43:29 PM Vallimamod Abdullah > wrote: > >> Hi, >> >> Is there any reason why you don't use the hash application directly >> instead of limit? >> IMHO it is more suited to your use case. >> >> https://freeswitch.org/confluence/display/FREESWITCH/mod_hash >> >> Best Regards, >> -- >> Vallimamod Abdullah >> SIP Solutions >> vma at sipsolutions.fr >> . >> >> >>> On 25 Jun 2017, at 17:10, Bipin Patel >> > wrote: >>> >>> hi, >>> >>> i have a requirement where i need to generate a different ringtone >>> if that ext is on another call and i do it using limit by setting a >>> counter for the caller and callee and then checking the counter if >>> it has a value of 1 which would mean that ext is on anohter call so >>> generate a different ringtone and this work fine. >>> >>> The problem being if A calls B and B attended transfers the call to >>> C then the counter for C is incremented in the features dialplan but >>> it doesnt reflect in the internal dialplan and i read that active >>> limits dont remain global across dialplans so is there anyway to >>> keep the limit counter active. i tried limit_ignore_transfer but >>> that doesnt seem to work, im on the FS master >>> >>> Below is my internal and features dialplan, any pointers would be >>> helpful >>> >>> >>> INTERNAL DIALPLAN >>> >>> >>> >>> >> data="rtp_negotiate_near_match=true"/> >>> >>> >>> >>> >>> >>> >>> >>> >>> >> expression="^ChangeRingBack$"/> >>> >>> >> data="transfer_ringback=%(250,250,400);%(250,3500,400)"/> >>> >> data="ringback=%(250,250,400);%(250,3500,400)"/> >>> >> data="sofia/internal/$1%$${domain}"/> >>> >>> >>> >>> >>> >>> >>> >>> >> data="transfer_ringback=%(400,200,400,375);%(400,2000,400,375)"/> >>> >>> >> data="ringback=%(400,200,400,375);%(400,2000,400,375)"/> >>> >>> >>> >>> >>> >> data="sofia/internal/$1%$${domain}"/> >>> >>> >>> >>> FEATURES DIALPLAN >>> >>> >>> >>> >> data="transfer_ringback=%(400,200,400,375);%(400,2000,400,375)"/> >>> >>> >> data="ringback=%(400,200,400,375);%(400,2000,400,375)"/> >>> >>> >>> >>> >>> >> data="sofia/internal/${digits}%$${domain}"/> >>> >>> >>> >>> >>> -- >>> Regards, >>> Bipin >>> >>> >>> ------------------------------------------------------------------------ >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at 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 at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From nbhatti at gmail.com Mon Jun 26 14:53:38 2017 From: nbhatti at gmail.com (Muhammad Naseer Bhatti) Date: Mon, 26 Jun 2017 07:53:38 -0700 Subject: [Freeswitch-users] FreeSWITCH support for RFC 4733 Message-ID: Hi, like the subject says, do we have support for RFC 4733? Can’t seem to find anything references in the code so far. -- Sent with Airmail -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Mon Jun 26 15:01:23 2017 From: mike at jerris.com (Michael Jerris) Date: Mon, 26 Jun 2017 11:01:23 -0400 Subject: [Freeswitch-users] FreeSWITCH support for RFC 4733 In-Reply-To: References: Message-ID: we have 2833… 4733 is mostly a super set of that. I have personally never seen a device that supports anything in the superset part, so we’ve never made any efforts to add any of the new tone stuff it can do, but regular dtmf handling interpos properly with almost all things out there. > On Jun 26, 2017, at 10:53 AM, Muhammad Naseer Bhatti wrote: > > Hi, like the subject says, do we have support for RFC 4733? Can’t seem to find anything references in the code so far. -------------- next part -------------- An HTML attachment was scrubbed... URL: From deepikay at iiitd.ac.in Mon Jun 26 15:14:26 2017 From: deepikay at iiitd.ac.in (Deepika Yadav) Date: Mon, 26 Jun 2017 20:44:26 +0530 Subject: [Freeswitch-users] Freeswitch API access In-Reply-To: References: Message-ID: Yeah sure, I would be happy to participate and get in touch with the community. Regards. Deepika On Sat, Jun 24, 2017 at 12:29 AM, Giovanni Maruzzelli wrote: > > > On 23 June 2017 at 19:16, Deepika Yadav wrote: > >> There is one more 'I' in the institute name IIIT (https://iiitd.ac.in) >> :). In my PhD work which belongs to the domain of HCI, broadly, I address >> the challenge of training Community Health Workers, who work in rural parts >> of India using mobile technology. I actively use Freeswitch in my work. I >> have built a platform that combines IVR and android app to provide a >> synchronous discussion forum for these health workers. Its use can be seen >> in two of the research papers: >> >> http://dl.acm.org/citation.cfm?id=3052624&CFID=777518217&CFTOKEN=57957032 >> >> http://dl.acm.org/citation.cfm?id=2858585 >> >> I find this Freedomfone software very interesting and will definitely >> explore it for our use cases. >> >> Again, thanks for sharing the information. >> > > Thanks to you Deepika, we're proud and glad FreeSWITCH is used in such a > project, and we would like so much to have you presenting it in one of our > weekly videoconferences, that are also on youtube. > > Please, be in contact and write to kathleen at freeswitch.org for our weekly > call. > > -giovanni > > > > >> >> Regards, >> Deepika >> >> On Fri, Jun 23, 2017 at 10:05 PM, Giovanni Maruzzelli >> wrote: >> >>> >>> >>> On 23 June 2017 at 18:30, Deepika Yadav wrote: >>> >>>> Thanks for the information. >>>> >>> >>> btw, Deepika, I had a look at your web page at IIT (I gave a little >>> conference there many years ago, iirc). Maybe you can find the >>> "Freedomfone" project interesting... http://freedomfone.org/ >>> >>> Is probably a little old but... >>> >>> Best wishes for your PHD, and please let us know how it goes, etc >>> >>> -giovanni >>> >>> >>> >>>> >>>> Regards, >>>> Deepika >>>> >>>> On Fri, Jun 23, 2017 at 9:55 PM, Giovanni Maruzzelli >>> > wrote: >>>> >>>>> >>>>> >>>>> On 23 June 2017 at 18:19, Deepika Yadav wrote: >>>>> >>>>>> Here the locks are simple Python locks to safeguard the execution of >>>>>> API. >>>>>> >>>>>> On Fri, Jun 23, 2017 at 9:48 PM, Deepika Yadav >>>>>> wrote: >>>>>> >>>>>>> Yes, there is python based server that receives requests from >>>>>>> Android apps. Based on a particular request, it initiates a conference call >>>>>>> among a set of users over cellular network. As soon as a new request comes, >>>>>>> the server starts a new thread that listens to Freeswitch events required >>>>>>> for taking some actions for this conference. >>>>>>> >>>>>>> I was just worried about the case where multiple requests come >>>>>>> making the server call multiple "originate API" with small gap leading to >>>>>>> some kind of problem at the Freeswitch side. >>>>>>> >>>>>> >>>>> >>>>> No problems at all. FreeSWITCH is a massively threaded server, each >>>>> thing is in its own thread, no need to serialize them at all. >>>>> >>>>> -giovanni >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>>> >>>>>>> Regards, >>>>>>> Deepika >>>>>>> >>>>>>> On Fri, Jun 23, 2017 at 9:37 PM, Giovanni Maruzzelli < >>>>>>> gmaruzz at gmail.com> wrote: >>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> On 23 June 2017 at 18:02, Deepika Yadav >>>>>>>> wrote: >>>>>>>> >>>>>>>>> Hi, >>>>>>>>> >>>>>>>>> I have a scenario where multiple conferences are established among >>>>>>>>> users. This is achieved through python programs over ESL. For each >>>>>>>>> conference, a new thread is created. So, should I also use locks before >>>>>>>>> firing originate calls APIs? >>>>>>>>> >>>>>>>> >>>>>>>> A new thread is created where by who? A new Python thread in your >>>>>>>> app? >>>>>>>> >>>>>>>> Which locks? On what? And why? >>>>>>>> >>>>>>>> You do not need any interprocess thing on FreeSWITCH side. FS will >>>>>>>> spawn internal threads to make outbound calls in parallel... >>>>>>>> >>>>>>>> -giovanni >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>>> >>>>>>>>> Regards >>>>>>>>> Deepika >>>>>>>>> https://www.iiitd.edu.in/~deepikay/ >>>>>>>>> >>>>>>>>> ____________________________________________________________ >>>>>>>>> _____________ >>>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>>> consulting at 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 at lists.freeswitch.org >>>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>>>>>>> switch-users >>>>>>>>> http://www.freeswitch.org >>>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> -- >>>>>>>> >>>>>>>> Sincerely, >>>>>>>> >>>>>>>> Giovanni Maruzzelli >>>>>>>> OpenTelecom.IT >>>>>>>> cell: +39 347 266 56 18 >>>>>>>> >>>>>>>> ____________________________________________________________ >>>>>>>> _____________ >>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>> consulting at 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 at lists.freeswitch.org >>>>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>>>>>> switch-users >>>>>>>> http://www.freeswitch.org >>>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> -- >>>>>>> Regards >>>>>>> Deepika >>>>>>> https://www.iiitd.edu.in/~deepikay/ >>>>>>> >>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> Regards >>>>>> Deepika >>>>>> https://www.iiitd.edu.in/~deepikay/ >>>>>> >>>>>> ____________________________________________________________ >>>>>> _____________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at 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 at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>>>> switch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>> >>>>> >>>>> >>>>> -- >>>>> >>>>> Sincerely, >>>>> >>>>> Giovanni Maruzzelli >>>>> OpenTelecom.IT >>>>> cell: +39 347 266 56 18 >>>>> >>>>> ____________________________________________________________ >>>>> _____________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at 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 at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>>> switch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> >>>> -- >>>> Regards >>>> Deepika >>>> https://www.iiitd.edu.in/~deepikay/ >>>> >>>> ____________________________________________________________ >>>> _____________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at 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 at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>> switch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> >>> -- >>> >>> Sincerely, >>> >>> Giovanni Maruzzelli >>> OpenTelecom.IT >>> cell: +39 347 266 56 18 >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at 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 at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Regards >> Deepika >> https://www.iiitd.edu.in/~deepikay/ >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at 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 > OpenTelecom.IT > cell: +39 347 266 56 18 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Regards Deepika https://www.iiitd.edu.in/~deepikay/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From admin at blindi.net Mon Jun 26 15:48:37 2017 From: admin at blindi.net (Thomas Hoellriegel) Date: Mon, 26 Jun 2017 17:48:37 +0200 (CEST) Subject: [Freeswitch-users] Information: German acoustical social network with fs Message-ID: Hi All, Fs is very nice to handle complex jobs. I have setup a german inclsionproject "dorf-telefonchat". This is a none commercial network. This is a virtual audioworld, with complex sounddesign, and ambiences. Facebook and other social networks are often not useable for handycap users, or users to having problems to get informations as barrierefree. The navigation is digits only (voicemenus). For examle: press 1 for originizer, 2, for messageboards 3 for mailbox on so on. I have setup sip, german landline, iax and skype, connections to call these system. Thanks. --------------- Website: www.dorf-telefonchat.de From gmaruzz at gmail.com Mon Jun 26 15:55:09 2017 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Mon, 26 Jun 2017 17:55:09 +0200 Subject: [Freeswitch-users] Information: German acoustical social network with fs In-Reply-To: References: Message-ID: On 26 June 2017 at 17:48, Thomas Hoellriegel wrote: > Hi All, > > Fs is very nice to handle complex jobs. > I have setup a german inclsionproject "dorf-telefonchat". > This is a none commercial network. > This is a virtual audioworld, with complex sounddesign, and ambiences. > Facebook and other social networks are often not useable for handycap > users, or users to having problems to get informations as barrierefree. > The navigation is digits only (voicemenus). For examle: > press 1 for originizer, 2, for messageboards 3 for mailbox on so on. > I have setup sip, german landline, iax and skype, connections to call > these system. > Seems very interesting, but website is in German, and most of us mailing readers cannot understand it. Can you please explain it more, and maybe present it to the community? We're very interested in this kinds of efforts. Thanks a lot for letting us know! -giovanni > > Thanks. > > > > > --------------- > Website: > www.dorf-telefonchat.de > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at 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 OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: From lconroy at insensate.co.uk Mon Jun 26 16:07:57 2017 From: lconroy at insensate.co.uk (Lawrence Conroy) Date: Mon, 26 Jun 2017 17:07:57 +0100 Subject: [Freeswitch-users] FreeSWITCH support for RFC 4733 In-Reply-To: References: Message-ID: Hi Michael, Naseer, folks, "It'll Work": Absolutely Agreed. Strictly, RFC2833 was kinda thrown together to capture what people knew or could guess then & to avoid everyone inventing their own incompatible schemes; Scott Petrack, plus Tom Taylor and his MEGACO crowd pulled together ideas on data that was carried/processed by kit as they had experienced/tested it AT THE TIME. After that, a number of interop issues (e.g., how long are those events?) occurred and RFC 4733 was intended to resolve those. Given two ways of interpreting things, some implementors will always invent their own new way :/ Some features had NOT really interworked reliably well (e.g., interpretation of a number of trunk & line events differed, depending on the gateway sending or receiving it), so those were deprecated. On the other hand, other hacks (or "neat ideas") had popped up in the mean time in different bits of kit, so there was a slight pressure to add those (generally pushed back). The doc itself was re-structured to make it a bit easier for mere mortals/implementers to understand. I'm not aware of any kit that claims 4733 compatibility that won't work with 2833 stacks; if there are, avoid! So ... apart from the process wonk bit on IANA registries, trying to re-write it in English & avoid ambiguities, it's pretty much the same as 2833. See Appendix A of for the changes. all the best Lawrence On 26 Jun 2017, at 16:01, Michael Jerris wrote: > we have 2833… 4733 is mostly a super set of that. I have personally never seen a device that supports anything in the superset part, so we’ve never made any efforts to add any of the new tone stuff it can do, but regular dtmf handling interpos properly with almost all things out there. > >> On Jun 26, 2017, at 10:53 AM, Muhammad Naseer Bhatti wrote: >> >> Hi, like the subject says, do we have support for RFC 4733? Can’t seem to find anything references in the code so far. From magnus.kelly at gmail.com Mon Jun 26 16:33:22 2017 From: magnus.kelly at gmail.com (Magnus) Date: Mon, 26 Jun 2017 17:33:22 +0100 Subject: [Freeswitch-users] Refer with replaces - how to configure to support In-Reply-To: References: Message-ID: Hello All, I had hoped that this would be straightforward in terms of asking if FS can support Refer with replaces support, but perhaps I'm asking in the wrong way, hence can anyone confirm that FS supports RFC3515, RFC3892, RFC589 ? I see all mentioned in FS specifications page, but yet to stumble into how it could/should be configured - Any/all tips welcome. Regards Magnus From anthony.minessale at gmail.com Mon Jun 26 19:37:56 2017 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 26 Jun 2017 14:37:56 -0500 Subject: [Freeswitch-users] The majority of ClueCon 2017 speakers have now been announced Message-ID: ClueCon is the annual celebration of FreeSWITCH and all other open source telephony. It is produced for all 13 years by the FreeSWITCH core team. Our motto is "a conference for developers, by developers" I hope we will see you there in August. Have a look at the lineup for 2017 on our website! https://www.cluecon.com/speakers.html -- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ * ClueCon Weekly Development Call ☎ sip:888 at conference.freeswitch.org ☎ +19193869900 https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA -------------- next part -------------- An HTML attachment was scrubbed... URL: From vma at vallimamod.org Tue Jun 27 06:58:13 2017 From: vma at vallimamod.org (Vallimamod Abdullah) Date: Tue, 27 Jun 2017 08:58:13 +0200 Subject: [Freeswitch-users] keep limit counter active across dialplans when used in attended transfers In-Reply-To: <752b7178-f635-a544-05a6-df985bfa851f@xbipin.com> References: <15ce0d1aa40.279b.b07ebdf329620b8089087c7205b03f01@xbipin.com> <752b7178-f635-a544-05a6-df985bfa851f@xbipin.com> Message-ID: <6418EDCA-2820-4C7B-B263-39A701FCF409@vallimamod.org> Hi, You can set hangup_after_bridge to false and delete the key after the bridge command. Best Regards, -- Vallimamod Abdullah SIP Solutions Consulting VOIP vma at sipsolutions.fr . > On 26 Jun 2017, at 15:59, Bipin Patel wrote: > > hi, > > i was testing mod_hash but how do we delete the key when the caller has hung up causing originator cancel because the dialplan doesnt execute any further, it does when the call is answered then hungup. > > Im assuming mod_hash works across dialplans because i didnt find anything in docs that would suggest it wont > > > Regards, > Bipin > > > -------- Original Message -------- > Subject: Re: [Freeswitch-users] keep limit counter active across dialplans when used in attended transfers > From: Bipin Patel > To: FreeSWITCH Users Help > Date: 6/25/2017, 11:54:16 PM >> Hi, >> >> Well I could use it but with limit the counter decrementing etc is automatic so it's much simpler and reduces the dialplan required. >> >> In the dialplan I provided how do you think I could use it directly without adding any more extensions in the two dialplan, any help would be appreciated. What I need to do is increment counter for caller and callee as well as callee transferred to and check that counter before bridging so change ring back can be called. >> >> On June 25, 2017 8:43:29 PM Vallimamod Abdullah wrote: >> >>> Hi, >>> >>> Is there any reason why you don't use the hash application directly instead of limit? >>> IMHO it is more suited to your use case. >>> >>> https://freeswitch.org/confluence/display/FREESWITCH/mod_hash >>> >>> Best Regards, >>> -- >>> Vallimamod Abdullah >>> SIP Solutions >>> vma at sipsolutions.fr >>> . >>> >>> >>>> On 25 Jun 2017, at 17:10, Bipin Patel > wrote: >>>> >>>> hi, >>>> >>>> i have a requirement where i need to generate a different ringtone if that ext is on another call and i do it using limit by setting a counter for the caller and callee and then checking the counter if it has a value of 1 which would mean that ext is on anohter call so generate a different ringtone and this work fine. >>>> >>>> The problem being if A calls B and B attended transfers the call to C then the counter for C is incremented in the features dialplan but it doesnt reflect in the internal dialplan and i read that active limits dont remain global across dialplans so is there anyway to keep the limit counter active. i tried limit_ignore_transfer but that doesnt seem to work, im on the FS master >>>> >>>> Below is my internal and features dialplan, any pointers would be helpful >>>> >>>> >>>> INTERNAL DIALPLAN >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> FEATURES DIALPLAN >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> -- >>>> Regards, >>>> Bipin >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at 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 at 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 at 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 at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From bipin at xbipin.com Tue Jun 27 07:06:03 2017 From: bipin at xbipin.com (Bipin Patel) Date: Tue, 27 Jun 2017 11:06:03 +0400 Subject: [Freeswitch-users] keep limit counter active across dialplans when used in attended transfers In-Reply-To: <6418EDCA-2820-4C7B-B263-39A701FCF409@vallimamod.org> References: <15ce0d1aa40.279b.b07ebdf329620b8089087c7205b03f01@xbipin.com> <752b7178-f635-a544-05a6-df985bfa851f@xbipin.com> <6418EDCA-2820-4C7B-B263-39A701FCF409@vallimamod.org> Message-ID: hi, yes i was trying that but the problem occurs when ur doing attended transfer because there when we set the hash key to oncall in the features dialplan, the bridge is actually a att_xfer so it doesnt run the delete command after C is hungup Regards, Bipin ------------------------------------------------------------------------ -------- Original Message -------- Subject: Re: [Freeswitch-users] keep limit counter active across dialplans when used in attended transfers From: Vallimamod Abdullah To: FreeSWITCH Users Help Date: 6/27/2017, 10:58:13 AM > Hi, > > You can set hangup_after_bridge to false and delete the key after the > bridge command. > > Best Regards, > -- > Vallimamod Abdullah > SIP Solutions > Consulting VOIP > vma at sipsolutions.fr > . > >> On 26 Jun 2017, at 15:59, Bipin Patel > > wrote: >> >> hi, >> >> i was testing mod_hash but how do we delete the key when the caller >> has hung up causing originator cancel because the dialplan doesnt >> execute any further, it does when the call is answered then hungup. >> >> Im assuming mod_hash works across dialplans because i didnt find >> anything in docs that would suggest it wont >> >> >> Regards, >> Bipin >> >> >> ------------------------------------------------------------------------ >> -------- Original Message -------- >> Subject: Re: [Freeswitch-users] keep limit counter active across >> dialplans when used in attended transfers >> From: Bipin Patel >> To: FreeSWITCH Users Help >> Date: 6/25/2017, 11:54:16 PM >>> >>> Hi, >>> >>> Well I could use it but with limit the counter decrementing etc is >>> automatic so it's much simpler and reduces the dialplan required. >>> >>> In the dialplan I provided how do you think I could use it directly >>> without adding any more extensions in the two dialplan, any help >>> would be appreciated. What I need to do is increment counter for >>> caller and callee as well as callee transferred to and check that >>> counter before bridging so change ring back can be called. >>> >>> On June 25, 2017 8:43:29 PM Vallimamod >>> Abdullahwrote: >>> >>>> Hi, >>>> >>>> Is there any reason why you don't use the hash application directly >>>> instead of limit? >>>> IMHO it is more suited to your use case. >>>> >>>> https://freeswitch.org/confluence/display/FREESWITCH/mod_hash >>>> >>>> Best Regards, >>>> -- >>>> Vallimamod Abdullah >>>> SIP Solutions >>>> vma at sipsolutions.fr >>>> . >>>> >>>> >>>>> On 25 Jun 2017, at 17:10, Bipin Patel >>>> > wrote: >>>>> >>>>> hi, >>>>> >>>>> i have a requirement where i need to generate a different ringtone >>>>> if that ext is on another call and i do it using limit by setting >>>>> a counter for the caller and callee and then checking the counter >>>>> if it has a value of 1 which would mean that ext is on anohter >>>>> call so generate a different ringtone and this work fine. >>>>> >>>>> The problem being if A calls B and B attended transfers the call >>>>> to C then the counter for C is incremented in the features >>>>> dialplan but it doesnt reflect in the internal dialplan and i read >>>>> that active limits dont remain global across dialplans so is there >>>>> anyway to keep the limit counter active. i tried >>>>> limit_ignore_transfer but that doesnt seem to work, im on the FS >>>>> master >>>>> >>>>> Below is my internal and features dialplan, any pointers would be >>>>> helpful >>>>> >>>>> >>>>> INTERNAL DIALPLAN >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>> data="transfer_ringback=%(250,250,400);%(250,3500,400)"/> >>>>> >>>> data="ringback=%(250,250,400);%(250,3500,400)"/> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>> data="transfer_ringback=%(400,200,400,375);%(400,2000,400,375)"/> >>>>> >>>>> >>>> data="ringback=%(400,200,400,375);%(400,2000,400,375)"/> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> FEATURES DIALPLAN >>>>> >>>>> >>>>> >>>>> >>>> data="transfer_ringback=%(400,200,400,375);%(400,2000,400,375)"/> >>>>> >>>>> >>>> data="ringback=%(400,200,400,375);%(400,2000,400,375)"/> >>>>> >>>>> >>>>> >>>>> >>>>> >>>> data="sofia/internal/${digits}%$${domain}"/> >>>>> >>>>> >>>>> >>>>> >>>>> -- >>>>> Regards, >>>>> Bipin >>>>> >>>>> >>>>> ------------------------------------------------------------------------ >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From bipin at xbipin.com Tue Jun 27 07:10:47 2017 From: bipin at xbipin.com (Bipin Patel) Date: Tue, 27 Jun 2017 11:10:47 +0400 Subject: [Freeswitch-users] keep limit counter active across dialplans when used in attended transfers In-Reply-To: References: <15ce0d1aa40.279b.b07ebdf329620b8089087c7205b03f01@xbipin.com> <752b7178-f635-a544-05a6-df985bfa851f@xbipin.com> <6418EDCA-2820-4C7B-B263-39A701FCF409@vallimamod.org> Message-ID: hi, and also hangup_after_bridge doesnt work to delete the key, need to use set_zombie_exec Regards, Bipin ------------------------------------------------------------------------ -------- Original Message -------- Subject: Re: [Freeswitch-users] keep limit counter active across dialplans when used in attended transfers From: Bipin Patel To: FreeSWITCH Users Help Date: 6/27/2017, 11:06:03 AM > hi, > > yes i was trying that but the problem occurs when ur doing attended > transfer because there when we set the hash key to oncall in the > features dialplan, the bridge is actually a att_xfer so it doesnt run > the delete command after C is hungup > > > Regards, > Bipin > > > ------------------------------------------------------------------------ > -------- Original Message -------- > Subject: Re: [Freeswitch-users] keep limit counter active across > dialplans when used in attended transfers > From: Vallimamod Abdullah > To: FreeSWITCH Users Help > Date: 6/27/2017, 10:58:13 AM >> Hi, >> >> You can set hangup_after_bridge to false and delete the key after the >> bridge command. >> >> Best Regards, >> -- >> Vallimamod Abdullah >> SIP Solutions >> Consulting VOIP >> vma at sipsolutions.fr >> . >> >>> On 26 Jun 2017, at 15:59, Bipin Patel >> > wrote: >>> >>> hi, >>> >>> i was testing mod_hash but how do we delete the key when the caller >>> has hung up causing originator cancel because the dialplan doesnt >>> execute any further, it does when the call is answered then hungup. >>> >>> Im assuming mod_hash works across dialplans because i didnt find >>> anything in docs that would suggest it wont >>> >>> >>> Regards, >>> Bipin >>> >>> >>> ------------------------------------------------------------------------ >>> -------- Original Message -------- >>> Subject: Re: [Freeswitch-users] keep limit counter active across >>> dialplans when used in attended transfers >>> From: Bipin Patel >>> To: FreeSWITCH Users Help >>> Date: 6/25/2017, 11:54:16 PM >>>> >>>> Hi, >>>> >>>> Well I could use it but with limit the counter decrementing etc is >>>> automatic so it's much simpler and reduces the dialplan required. >>>> >>>> In the dialplan I provided how do you think I could use it directly >>>> without adding any more extensions in the two dialplan, any help >>>> would be appreciated. What I need to do is increment counter for >>>> caller and callee as well as callee transferred to and check that >>>> counter before bridging so change ring back can be called. >>>> >>>> On June 25, 2017 8:43:29 PM Vallimamod >>>> Abdullahwrote: >>>> >>>>> Hi, >>>>> >>>>> Is there any reason why you don't use the hash application >>>>> directly instead of limit? >>>>> IMHO it is more suited to your use case. >>>>> >>>>> https://freeswitch.org/confluence/display/FREESWITCH/mod_hash >>>>> >>>>> Best Regards, >>>>> -- >>>>> Vallimamod Abdullah >>>>> SIP Solutions >>>>> vma at sipsolutions.fr >>>>> . >>>>> >>>>> >>>>>> On 25 Jun 2017, at 17:10, Bipin Patel >>>>> > wrote: >>>>>> >>>>>> hi, >>>>>> >>>>>> i have a requirement where i need to generate a different >>>>>> ringtone if that ext is on another call and i do it using limit >>>>>> by setting a counter for the caller and callee and then checking >>>>>> the counter if it has a value of 1 which would mean that ext is >>>>>> on anohter call so generate a different ringtone and this work fine. >>>>>> >>>>>> The problem being if A calls B and B attended transfers the call >>>>>> to C then the counter for C is incremented in the features >>>>>> dialplan but it doesnt reflect in the internal dialplan and i >>>>>> read that active limits dont remain global across dialplans so is >>>>>> there anyway to keep the limit counter active. i tried >>>>>> limit_ignore_transfer but that doesnt seem to work, im on the FS >>>>>> master >>>>>> >>>>>> Below is my internal and features dialplan, any pointers would be >>>>>> helpful >>>>>> >>>>>> >>>>>> INTERNAL DIALPLAN >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>> data="transfer_ringback=%(250,250,400);%(250,3500,400)"/> >>>>>> >>>>> data="ringback=%(250,250,400);%(250,3500,400)"/> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>> data="transfer_ringback=%(400,200,400,375);%(400,2000,400,375)"/> >>>>>> >>>>>> >>>>> data="ringback=%(400,200,400,375);%(400,2000,400,375)"/> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> FEATURES DIALPLAN >>>>>> >>>>>> >>>>>> >>>>>> >>>>> data="transfer_ringback=%(400,200,400,375);%(400,2000,400,375)"/> >>>>>> >>>>>> >>>>> data="ringback=%(400,200,400,375);%(400,2000,400,375)"/> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>> data="sofia/internal/${digits}%$${domain}"/> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> -- >>>>>> Regards, >>>>>> Bipin >>>>>> >>>>>> >>>>>> ------------------------------------------------------------------------ >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From igorolhovskiy at gmail.com Tue Jun 27 08:46:02 2017 From: igorolhovskiy at gmail.com (Igor Olhovskiy) Date: Tue, 27 Jun 2017 11:46:02 +0300 Subject: [Freeswitch-users] Recording distorted on resume Message-ID: <55b70971-3b86-4332-bc1d-d08da2be939f@Spark> Hi! I’m trying to make pause in recording via bind_digit_action record_session and stop_record_session RECORD_APPEND is set to true, format is WAV Problem I have is when I’m listening to recording, first part is distorted completely (sounds like 44kHz is transformed to 8kHz with slowdown and not with resampling) Recording is done via export(nolocal:api_on_answer=uuid_record 5ae06306-70a5-4e0b-a9ca-cb44ebf0f95b start /var/lib/freeswitch/recordings/bw.1st-byte.com/archive/2017/Jun/27/5ae06306-70a5-4e0b-a9ca-cb44ebf0f95b.wav) Than bind_digit_action(local,*2,exec:record_session,/var/lib/freeswitch/recordings/bw.1st-byte.com/archive/2017/Jun/27/5ae06306-70a5-4e0b-a9ca-cb44ebf0f95b.wav,both) and bind_digit_action(local,*1,exec:stop_record_session,/var/lib/freeswitch/recordings/bw.1st-byte.com/archive/2017/Jun/27/5ae06306-70a5-4e0b-a9ca-cb44ebf0f95b.wav,both) This is a full log of a call https://pastebin.com/4Nn7dhui Freeswitch version is 1.6.18-35-6e79667~64bit Regards, Igor -------------- next part -------------- An HTML attachment was scrubbed... URL: From igorolhovskiy at gmail.com Tue Jun 27 10:26:55 2017 From: igorolhovskiy at gmail.com (Igor Olhovskiy) Date: Tue, 27 Jun 2017 13:26:55 +0300 Subject: [Freeswitch-users] Recording distorted on resume In-Reply-To: <55b70971-3b86-4332-bc1d-d08da2be939f@Spark> References: <55b70971-3b86-4332-bc1d-d08da2be939f@Spark> Message-ID: <3e7d3f5c-f4a9-4d98-8d48-e0e3ed5b3c69@Spark> After playing a while (replace api_on_answer with execute_on_answer) got situation when 2nd part of conversation (after a pause) used as a first part also. So, I have 2 second parts in conversation. Quite strange. Regards, Igor On 27 июня 2017 г., 11:51 +0300, Igor Olhovskiy , wrote: > Hi! > > I’m trying to make pause in recording via > bind_digit_action record_session and stop_record_session > RECORD_APPEND is set to true, format is WAV > > Problem I have is when I’m listening to recording, first part is distorted completely (sounds like 44kHz is transformed to 8kHz with slowdown and not with resampling) > > Recording is done via > > export(nolocal:api_on_answer=uuid_record 5ae06306-70a5-4e0b-a9ca-cb44ebf0f95b start /var/lib/freeswitch/recordings/bw.1st-byte.com/archive/2017/Jun/27/5ae06306-70a5-4e0b-a9ca-cb44ebf0f95b.wav) > > Than bind_digit_action(local,*2,exec:record_session,/var/lib/freeswitch/recordings/bw.1st-byte.com/archive/2017/Jun/27/5ae06306-70a5-4e0b-a9ca-cb44ebf0f95b.wav,both) > > and > bind_digit_action(local,*1,exec:stop_record_session,/var/lib/freeswitch/recordings/bw.1st-byte.com/archive/2017/Jun/27/5ae06306-70a5-4e0b-a9ca-cb44ebf0f95b.wav,both) > > This is a full log of a call > https://pastebin.com/4Nn7dhui > > Freeswitch version is 1.6.18-35-6e79667~64bit > > > Regards, Igor -------------- next part -------------- An HTML attachment was scrubbed... URL: From igorolhovskiy at gmail.com Tue Jun 27 10:52:34 2017 From: igorolhovskiy at gmail.com (Igor Olhovskiy) Date: Tue, 27 Jun 2017 13:52:34 +0300 Subject: [Freeswitch-users] Recording distorted on resume In-Reply-To: <55b70971-3b86-4332-bc1d-d08da2be939f@Spark> References: <55b70971-3b86-4332-bc1d-d08da2be939f@Spark> Message-ID: Ok, I’ve resolved this in other way, but seems, it’s a bug in freeswitch anyway. Workaround - use uuid_record mask/unmask Regards, Igor On 27 июня 2017 г., 11:51 +0300, Igor Olhovskiy , wrote: > Hi! > > I’m trying to make pause in recording via > bind_digit_action record_session and stop_record_session > RECORD_APPEND is set to true, format is WAV > > Problem I have is when I’m listening to recording, first part is distorted completely (sounds like 44kHz is transformed to 8kHz with slowdown and not with resampling) > > Recording is done via > > export(nolocal:api_on_answer=uuid_record 5ae06306-70a5-4e0b-a9ca-cb44ebf0f95b start /var/lib/freeswitch/recordings/bw.1st-byte.com/archive/2017/Jun/27/5ae06306-70a5-4e0b-a9ca-cb44ebf0f95b.wav) > > Than bind_digit_action(local,*2,exec:record_session,/var/lib/freeswitch/recordings/bw.1st-byte.com/archive/2017/Jun/27/5ae06306-70a5-4e0b-a9ca-cb44ebf0f95b.wav,both) > > and > bind_digit_action(local,*1,exec:stop_record_session,/var/lib/freeswitch/recordings/bw.1st-byte.com/archive/2017/Jun/27/5ae06306-70a5-4e0b-a9ca-cb44ebf0f95b.wav,both) > > This is a full log of a call > https://pastebin.com/4Nn7dhui > > Freeswitch version is 1.6.18-35-6e79667~64bit > > > Regards, Igor -------------- next part -------------- An HTML attachment was scrubbed... URL: From deepikay at iiitd.ac.in Tue Jun 27 11:06:10 2017 From: deepikay at iiitd.ac.in (Deepika Yadav) Date: Tue, 27 Jun 2017 16:36:10 +0530 Subject: [Freeswitch-users] VOIP-GSM Gateway configuration Error Message-ID: Hi, I have been trying for a while the configuration of a VOIP-GSM gateway that I have bought but the registration fails every time. Error at Freeswitch console 2017-06-27 16:27:10.191913 [WARNING] sofia_reg.c:2906 Can't find user [*testadmin at 192.168.2.71 *] from *192.168.2.69* You must define a domain called '192.168.2.71' in your directory and add a user with the id="*testadmin*" attribute and you must configure your device to use the proper domain in it's authentication credentials. 2017-06-27 16:27:14.071916 [ERR] sofia_reg.c:2447 MySIP Failed Registration with status Request Timeout [408]. failure #1 2017-06-27 16:27:14.111913 [WARNING] sofia_reg.c:505 MySIP Failed Registration [408], setting retry to 30 seconds. Here *'192.168.2.71*' is the local IP of the Freeswitch server and *192.168.2.69* that of the gateway attached to the LAN port of the Freeswitch server. I have put SIP UserID in the gateway admin panel as "*testadmin*" alongwith its passoword. my external SIP profile is as follows: I have tried number of permutations and combinations but the gateway is not getting registered. Regards Deepika https://www.iiitd.edu.in/~deepikay/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From igorolhovskiy at gmail.com Tue Jun 27 11:14:40 2017 From: igorolhovskiy at gmail.com (Igor Olhovskiy) Date: Tue, 27 Jun 2017 14:14:40 +0300 Subject: [Freeswitch-users] VOIP-GSM Gateway configuration Error In-Reply-To: References: Message-ID: <238564c8-4d54-4354-b810-2340514c8902@Spark> «gateway" entity in Freeswitch is supposed to register on external services. If you can make Freeswitch register on your VOIP-GSM gateway - this may resolve your issue Regards, Igor On 27 июня 2017 г., 14:07 +0300, Deepika Yadav , wrote: > Hi, > > I have been trying for a while the configuration of a VOIP-GSM gateway that I have bought but the registration fails every time. > > Error at Freeswitch console > > 2017-06-27 16:27:10.191913 [WARNING] sofia_reg.c:2906 Can't find user [testadmin at 192.168.2.71] from 192.168.2.69 > You must define a domain called '192.168.2.71' in your directory and add a user with the id="testadmin" attribute > and you must configure your device to use the proper domain in it's authentication credentials. > 2017-06-27 16:27:14.071916 [ERR] sofia_reg.c:2447 MySIP Failed Registration with status Request Timeout [408]. failure #1 > 2017-06-27 16:27:14.111913 [WARNING] sofia_reg.c:505 MySIP Failed Registration [408], setting retry to 30 seconds. > > Here '192.168.2.71' is the local IP of the Freeswitch server and 192.168.2.69 that of the gateway attached to the LAN port of the Freeswitch server. > > I have put SIP UserID in the gateway admin panel as "testadmin" alongwith its passoword. > > my external SIP profile is as follows: > > > > > >   > >   > >   > >   > >   > >   > >   > >   > >   > >   > > > I have tried number of permutations and combinations but the gateway is not getting registered. > > > Regards > Deepika > https://www.iiitd.edu.in/~deepikay/ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From deepikay at iiitd.ac.in Tue Jun 27 11:33:28 2017 From: deepikay at iiitd.ac.in (Deepika Yadav) Date: Tue, 27 Jun 2017 17:03:28 +0530 Subject: [Freeswitch-users] VOIP-GSM Gateway configuration Error In-Reply-To: <238564c8-4d54-4354-b810-2340514c8902@Spark> References: <238564c8-4d54-4354-b810-2340514c8902@Spark> Message-ID: Sorry didn't get you, which gateway entity, where? This sip external profile is for adding the details of the VOIP-GSM gateway only, earlier I was using this service from a company and the configuration worked fine. But somehow, the gateway attached locally to the machine is not getting configured properly. Regards, Deepika On Tue, Jun 27, 2017 at 4:44 PM, Igor Olhovskiy wrote: > «gateway" entity in Freeswitch is supposed to register on external > services. > > If you can make Freeswitch register on your VOIP-GSM gateway - this may > resolve your issue > > Regards, Igor > > On 27 июня 2017 г., 14:07 +0300, Deepika Yadav , > wrote: > > Hi, > > I have been trying for a while the configuration of a VOIP-GSM gateway > that I have bought but the registration fails every time. > > Error at Freeswitch console > > 2017-06-27 16:27:10.191913 [WARNING] sofia_reg.c:2906 Can't find user [*testadmin at 192.168.2.71 > *] from *192.168.2.69* > You must define a domain called '192.168.2.71' in your directory and add a > user with the id="*testadmin*" attribute > and you must configure your device to use the proper domain in it's > authentication credentials. > 2017-06-27 16:27:14.071916 [ERR] sofia_reg.c:2447 MySIP Failed > Registration with status Request Timeout [408]. failure #1 > 2017-06-27 16:27:14.111913 [WARNING] sofia_reg.c:505 MySIP Failed > Registration [408], setting retry to 30 seconds. > > Here *'192.168.2.71*' is the local IP of the Freeswitch server and > *192.168.2.69* that of the gateway attached to the LAN port of the > Freeswitch server. > > I have put SIP UserID in the gateway admin panel as "*testadmin*" > alongwith its passoword. > > my external SIP profile is as follows: > > > > > > > > > > > > > > > > > > > > > > > > > > > I have tried number of permutations and combinations but the gateway is > not getting registered. > > > Regards > Deepika > https://www.iiitd.edu.in/~deepikay/ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Regards Deepika https://www.iiitd.edu.in/~deepikay/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From david.villasmil.work at gmail.com Tue Jun 27 11:46:31 2017 From: david.villasmil.work at gmail.com (David Villasmil) Date: Tue, 27 Jun 2017 11:46:31 +0000 Subject: [Freeswitch-users] VOIP-GSM Gateway configuration Error In-Reply-To: References: <238564c8-4d54-4354-b810-2340514c8902@Spark> Message-ID: Would be much simpler of you shows us what parameters are available on the voip-gsm. On Tue, Jun 27, 2017 at 1:34 PM Deepika Yadav wrote: > Sorry didn't get you, which gateway entity, where? > > This sip external profile is for adding the details of the VOIP-GSM > gateway only, earlier I was using this service from a company and the > configuration worked fine. But somehow, the gateway attached locally to the > machine is not getting configured properly. > > Regards, > Deepika > > On Tue, Jun 27, 2017 at 4:44 PM, Igor Olhovskiy > wrote: > >> «gateway" entity in Freeswitch is supposed to register on external >> services. >> >> If you can make Freeswitch register on your VOIP-GSM gateway - this may >> resolve your issue >> >> Regards, Igor >> >> On 27 июня 2017 г., 14:07 +0300, Deepika Yadav , >> wrote: >> >> Hi, >> >> I have been trying for a while the configuration of a VOIP-GSM gateway >> that I have bought but the registration fails every time. >> >> Error at Freeswitch console >> >> 2017-06-27 16:27:10.191913 [WARNING] sofia_reg.c:2906 Can't find user [*testadmin at 192.168.2.71 >> *] from *192.168.2.69* >> You must define a domain called '192.168.2.71' in your directory and add >> a user with the id="*testadmin*" attribute >> and you must configure your device to use the proper domain in it's >> authentication credentials. >> 2017-06-27 16:27:14.071916 [ERR] sofia_reg.c:2447 MySIP Failed >> Registration with status Request Timeout [408]. failure #1 >> 2017-06-27 16:27:14.111913 [WARNING] sofia_reg.c:505 MySIP Failed >> Registration [408], setting retry to 30 seconds. >> >> Here *'192.168.2.71*' is the local IP of the Freeswitch server and >> *192.168.2.69* that of the gateway attached to the LAN port of the >> Freeswitch server. >> >> I have put SIP UserID in the gateway admin panel as "*testadmin*" >> alongwith its passoword. >> >> my external SIP profile is as follows: >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> I have tried number of permutations and combinations but the gateway is >> not getting registered. >> >> >> Regards >> Deepika >> https://www.iiitd.edu.in/~deepikay/ >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at 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 at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Regards > Deepika > https://www.iiitd.edu.in/~deepikay/ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From igorolhovskiy at gmail.com Tue Jun 27 11:57:15 2017 From: igorolhovskiy at gmail.com (Igor Olhovskiy) Date: Tue, 27 Jun 2017 14:57:15 +0300 Subject: [Freeswitch-users] VOIP-GSM Gateway configuration Error In-Reply-To: References: <238564c8-4d54-4354-b810-2340514c8902@Spark> Message-ID: «Gateway» means https://freeswitch.org/confluence/display/FREESWITCH/Gateways+Configuration If you register with freeswtich on external services - you use gateway If you register some endpoints on freeswitch (which you are doing, trying to register your voip-gsm gateway) - it’s not a gateway, it’s extension. You can find how to dial with registered users here: https://freeswitchforum.com/viewtopic.php?f=6&t=312#p1466 It’s on russian, but you can use google translate Regards, Igor On 27 июня 2017 г., 14:47 +0300, David Villasmil , wrote: > Would be much simpler of you shows us what parameters are available on the voip-gsm. > > > On Tue, Jun 27, 2017 at 1:34 PM Deepika Yadav wrote: > > > Sorry didn't get you, which gateway entity, where? > > > > > > This sip external profile is for adding the details of the VOIP-GSM gateway only, earlier I was using this service from a company and the configuration worked fine. But somehow, the gateway attached locally to the machine is not getting configured properly. > > > > > > Regards, > > > Deepika > > > > > > > On Tue, Jun 27, 2017 at 4:44 PM, Igor Olhovskiy wrote: > > > > > «gateway" entity in Freeswitch is supposed to register on external services. > > > > > > > > > > If you can make Freeswitch register on your VOIP-GSM gateway - this may resolve your issue > > > > > > > > > > Regards, Igor > > > > > > > > > > On 27 июня 2017 г., 14:07 +0300, Deepika Yadav , wrote: > > > > > > Hi, > > > > > > > > > > > > I have been trying for a while the configuration of a VOIP-GSM gateway that I have bought but the registration fails every time. > > > > > > > > > > > > Error at Freeswitch console > > > > > > > > > > > > 2017-06-27 16:27:10.191913 [WARNING] sofia_reg.c:2906 Can't find user [testadmin at 192.168.2.71] from 192.168.2.69 > > > > > > You must define a domain called '192.168.2.71' in your directory and add a user with the id="testadmin" attribute > > > > > > and you must configure your device to use the proper domain in it's authentication credentials. > > > > > > 2017-06-27 16:27:14.071916 [ERR] sofia_reg.c:2447 MySIP Failed Registration with status Request Timeout [408]. failure #1 > > > > > > 2017-06-27 16:27:14.111913 [WARNING] sofia_reg.c:505 MySIP Failed Registration [408], setting retry to 30 seconds. > > > > > > > > > > > > Here '192.168.2.71' is the local IP of the Freeswitch server and 192.168.2.69 that of the gateway attached to the LAN port of the Freeswitch server. > > > > > > > > > > > > I have put SIP UserID in the gateway admin panel as "testadmin" alongwith its passoword. > > > > > > > > > > > > my external SIP profile is as follows: > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > >   > > > > > > > > > > > >   > > > > > > > > > > > >   > > > > > > > > > > > >   > > > > > > > > > > > >   > > > > > > > > > > > >   > > > > > > > > > > > >   > > > > > > > > > > > >   > > > > > > > > > > > >   > > > > > > > > > > > >   > > > > > > > > > > > > > > > > > > I have tried number of permutations and combinations but the gateway is not getting registered. > > > > > > > > > > > > > > > > > > Regards > > > > > > Deepika > > > > > > https://www.iiitd.edu.in/~deepikay/ > > > > > > _________________________________________________________________________ > > > > > > Professional FreeSWITCH Consulting Services: > > > > > > consulting at 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 at 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 at 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 at lists.freeswitch.org > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > > http://www.freeswitch.org > > > > > > > > > > > > -- > > > Regards > > > Deepika > > > https://www.iiitd.edu.in/~deepikay/ > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From covici at ccs.covici.com Tue Jun 27 12:19:17 2017 From: covici at ccs.covici.com (John Covici) Date: Tue, 27 Jun 2017 08:19:17 -0400 Subject: [Freeswitch-users] Recording distorted on resume In-Reply-To: References: <55b70971-3b86-4332-bc1d-d08da2be939f@Spark> Message-ID: I just stop the recording and start it again with the append flag set, works for me all the time. On Tue, 27 Jun 2017 06:52:34 -0400, Igor Olhovskiy wrote: > > [1 ] > [1.1 ] > [1.2 ] > Ok, I’ve resolved this in other way, but seems, it’s a bug in freeswitch anyway. > > Workaround - use uuid_record mask/unmask > > Regards, Igor > > On 27 июня 2017 г., 11:51 +0300, Igor Olhovskiy , wrote: > > Hi! > > I’m trying to make pause in recording via > bind_digit_action record_session and stop_record_session > RECORD_APPEND is set to true, format is WAV > > Problem I have is when I’m listening to recording, first part is distorted completely (sounds like 44kHz is transformed to 8kHz with slowdown and not with resampling) > > Recording is done via > > export(nolocal:api_on_answer=uuid_record 5ae06306-70a5-4e0b-a9ca-cb44ebf0f95b start /var/lib/freeswitch/recordings/bw.1st-byte.com/archive/2017/Jun/27/5ae06306-70a5-4e0b-a9ca-cb44ebf0f95b.wav) > > Than bind_digit_action(local,*2,exec:record_session,/var/lib/freeswitch/recordings/bw.1st-byte.com/archive/2017/Jun/27/5ae06306-70a5-4e0b-a9ca-cb44ebf0f95b.wav,both) > > and > bind_digit_action(local,*1,exec:stop_record_session,/var/lib/freeswitch/recordings/bw.1st-byte.com/archive/2017/Jun/27/5ae06306-70a5-4e0b-a9ca-cb44ebf0f95b.wav,both) > > This is a full log of a call > https://pastebin.com/4Nn7dhui > > Freeswitch version is 1.6.18-35-6e79667~64bit > > Regards, Igor > > [2 ] > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici at ccs.covici.com From italo at freeswitch.org Tue Jun 27 15:21:29 2017 From: italo at freeswitch.org (=?UTF-8?B?w410YWxvIFJvc3Np?=) Date: Tue, 27 Jun 2017 12:21:29 -0300 Subject: [Freeswitch-users] Recording distorted on resume In-Reply-To: References: <55b70971-3b86-4332-bc1d-d08da2be939f@Spark> Message-ID: If you think it's a bug please open a JIRA for it.. On Tue, Jun 27, 2017 at 9:19 AM, John Covici wrote: > I just stop the recording and start it again with the append flag set, > works for me all the time. > > On Tue, 27 Jun 2017 06:52:34 -0400, > Igor Olhovskiy wrote: > > > > [1 ] > > [1.1 ] > > [1.2 ] > > Ok, I’ve resolved this in other way, but seems, it’s a bug in freeswitch > anyway. > > > > Workaround - use uuid_record mask/unmask > > > > Regards, Igor > > > > On 27 июня 2017 г., 11:51 +0300, Igor Olhovskiy , > wrote: > > > > Hi! > > > > I’m trying to make pause in recording via > > bind_digit_action record_session and stop_record_session > > RECORD_APPEND is set to true, format is WAV > > > > Problem I have is when I’m listening to recording, first part is > distorted completely (sounds like 44kHz is transformed to 8kHz with > slowdown and not with resampling) > > > > Recording is done via > > > > export(nolocal:api_on_answer=uuid_record 5ae06306-70a5-4e0b-a9ca-cb44ebf0f95b > start /var/lib/freeswitch/recordings/bw.1st-byte.com/ > archive/2017/Jun/27/5ae06306-70a5-4e0b-a9ca-cb44ebf0f95b.wav) > > > > Than bind_digit_action(local,*2,exec:record_session,/var/lib/ > freeswitch/recordings/bw.1st-byte.com/archive/2017/Jun/27/ > 5ae06306-70a5-4e0b-a9ca-cb44ebf0f95b.wav,both) > > > > and > > bind_digit_action(local,*1,exec:stop_record_session,/var/ > lib/freeswitch/recordings/bw.1st-byte.com/archive/2017/Jun/ > 27/5ae06306-70a5-4e0b-a9ca-cb44ebf0f95b.wav,both) > > > > This is a full log of a call > > https://pastebin.com/4Nn7dhui > > > > Freeswitch version is 1.6.18-35-6e79667~64bit > > > > Regards, Igor > > > > [2 ] > > ____________________________________________________________ > _____________ > > Professional FreeSWITCH Consulting Services: > > consulting at 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 at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > -- > Your life is like a penny. You're going to lose it. The question is: > How do > you spend it? > > John Covici > covici at ccs.covici.com > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ítalo Rossi italo at freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From nbhatti at gmail.com Tue Jun 27 18:10:22 2017 From: nbhatti at gmail.com (Muhammad Naseer Bhatti) Date: Tue, 27 Jun 2017 20:10:22 +0200 Subject: [Freeswitch-users] How to delay setInputCallback() while playing audio? Message-ID: I am running an IVR which collects DTMF input while the file is being played, written in Lua. I set InputCallback and then listen for DTMF events. If event found, take actions etc. session:setInputCallback("eventHandler") playbackMessage (playback_delay) Facing a strange issues with Verizon where they send a quick RTP Event write after starting the audio, which triggers the DTMF event and bad things happen in my script. This only happens within first 1 to 2 seconds of RTP. How can I have my InputCallback listen for events only after a few seconds while still playing the audio? or do I have to switch to playAndGetDigits() ? -- Sent with Airmail -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian at freeswitch.org Tue Jun 27 19:12:36 2017 From: brian at freeswitch.org (Brian West) Date: Tue, 27 Jun 2017 14:12:36 -0500 Subject: [Freeswitch-users] How to delay setInputCallback() while playing audio? In-Reply-To: References: Message-ID: Show us how you're doing the playback, If you're doing session:execute('playback', 'blah.wav'); The input callback won't function you have to use session:streamFile instead. /b On Tue, Jun 27, 2017 at 1:10 PM, Muhammad Naseer Bhatti wrote: > I am running an IVR which collects DTMF input while the file is being > played, written in Lua. I set InputCallback and then listen for DTMF > events. If event found, take actions etc. > > session:setInputCallback("eventHandler") > playbackMessage (playback_delay) > > Facing a strange issues with Verizon where they send a quick RTP Event > write after starting the audio, which triggers the DTMF event and bad > things happen in my script. This only happens within first 1 to 2 seconds > of RTP. How can I have my InputCallback listen for events only after a few > seconds while still playing the audio? or do I have to switch to > playAndGetDigits() ? > > > -- > > Sent with Airmail > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Book a phone call (CST) Allison prompts for FreeSWITCH: *https://www.gofundme.com/allison-prompts-for-freeswitch* Got Bugs? Report them here ! | Reddit: /r/freeswitch *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: From nbhatti at gmail.com Tue Jun 27 19:18:40 2017 From: nbhatti at gmail.com (Muhammad Naseer Bhatti) Date: Tue, 27 Jun 2017 12:18:40 -0700 Subject: [Freeswitch-users] How to delay setInputCallback() while playing audio? In-Reply-To: References: Message-ID: playbackMessage() function is actually doing session:streamFile(audio_file). Input callback is functioning fine, but I want to delay detection a few seconds somehow. On the other hand, is there a way to stop a running session:streamFile() ? -- Sent with Airmail From: Brian West Reply: FreeSWITCH Users Help Date: June 28, 2017 at 12:15:20 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to delay setInputCallback() while playing audio? Show us how you're doing the playback, If you're doing session:execute('playback', 'blah.wav'); The input callback won't function you have to use session:streamFile instead. /b On Tue, Jun 27, 2017 at 1:10 PM, Muhammad Naseer Bhatti wrote: > I am running an IVR which collects DTMF input while the file is being > played, written in Lua. I set InputCallback and then listen for DTMF > events. If event found, take actions etc. > > session:setInputCallback("eventHandler") > playbackMessage (playback_delay) > > Facing a strange issues with Verizon where they send a quick RTP Event > write after starting the audio, which triggers the DTMF event and bad > things happen in my script. This only happens within first 1 to 2 seconds > of RTP. How can I have my InputCallback listen for events only after a few > seconds while still playing the audio? or do I have to switch to > playAndGetDigits() ? > > > -- > > Sent with Airmail > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Book a phone call (CST) Allison prompts for FreeSWITCH: *https://www.gofundme.com/allison-prompts-for-freeswitch* Got Bugs? Report them here ! | Reddit: /r/freeswitch *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *Skype:*briankwest _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From grcamauer at gmail.com Tue Jun 27 21:37:10 2017 From: grcamauer at gmail.com (Guillermo Ruiz Camauer) Date: Tue, 27 Jun 2017 18:37:10 -0300 Subject: [Freeswitch-users] Problem with fs_encode In-Reply-To: References: Message-ID: Anthony, I wasn't recompiling the app, which I now managed to do and everything is back to normal. Thanks! So if I understand correctly, I must have a version of the app for my "old" FreeSwitch boxes and another one for my "up to date" FreeSwitch boxes. Correct? I guess it's time to update all boxes... Thanks again, Guillermo On Sat, Jun 24, 2017 at 12:57 AM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > Are you recompiling the app too against the new code. The patch changes > abi of the fs libs. > > > On Fri, Jun 23, 2017 at 6:35 PM Guillermo Ruiz Camauer < > grcamauer at gmail.com> wrote: > >> Anthony, >> >> >> Removing these lines: >> >> + int scale_w; >> + int scale_h; >> + switch_img_fmt_t fmt; >> + char *auth_username; >> + char *auth_password; >> >> from switch_module_interfaces.h >> >> makes my program work again. Any idea of what they could be affecting? >> >> Guillermo >> >> On Fri, Jun 23, 2017 at 8:16 PM, Anthony Minessale < >> anthony.minessale at gmail.com> wrote: >> >>> My advise is to verify by removing bits or even all of the patch and >>> double check you did not mess up on the bisect. >>> like if you can manually reproduce going to the version before this >>> patch and having it work then go to the version with the patch and try >>> again etc. >>> >>> >>> >>> On Fri, Jun 23, 2017 at 6:14 PM, Guillermo Ruiz Camauer < >>> grcamauer at gmail.com> wrote: >>> >>>> I will try removing just those lines in switch_core_file.c. Maybe >>>> it's one of the other changes, I was just guessing... >>>> >>>> Guillermo >>>> >>>> On Fri, Jun 23, 2017 at 8:03 PM, Anthony Minessale < >>>> anthony.minessale at gmail.com> wrote: >>>> >>>>> Also does removing just that change in switch_core_file.c from latest >>>>> version actually make it work? It seems unlikely the next line in the file >>>>> looking for FPS field is no different really. >>>>> >>>>> >>>>> >>>>> >>>>> On Fri, Jun 23, 2017 at 6:01 PM, Michael Jerris >>>>> wrote: >>>>> >>>>>> can you drop a core file and see where its stuck? >>>>>> >>>>>> On Jun 23, 2017, at 6:57 PM, Guillermo Ruiz Camauer < >>>>>> grcamauer at gmail.com> wrote: >>>>>> >>>>>> I have created a small utility program based on fs_encode which will >>>>>> take ALL files in a directory and convert them from .WAV to .G729 and .PCMU >>>>>> formats. The utility is called wavBatchEncode and takes just a directory >>>>>> path as an argument. >>>>>> >>>>>> This program was working very well until I upgraded FreeSwitch. Now >>>>>> the program seems to hang when it tries to write out the first converted >>>>>> files. I have to KILL the program from another terminal. It leaves 2 0 >>>>>> byte files with .PCMU and .G729 extension. >>>>>> >>>>>> I have run through a GIT bisect to find where things broke. The >>>>>> last working version is Version 1.6.16 git ae1cdce 2017-04-11. >>>>>> The first broken version is 38621e47bad3b63f03a0a27f6ca9ed92f6969032. >>>>>> >>>>>> I then used GIT DIFF to see if I could see what had happened. There >>>>>> are very few modifications between these two commits: >>>>>> >>>>>> root at fs3:/usr/src/freeswitch.git# git diff ae1cdce >>>>>> 38621e47bad3b63f03a0a27f6ca9ed92f6969032 >>>>>> diff --git a/src/include/switch_module_interfaces.h >>>>>> b/src/include/switch_module_interfaces.h >>>>>> index e0a5c20..7ca027d 100644 >>>>>> --- a/src/include/switch_module_interfaces.h >>>>>> +++ b/src/include/switch_module_interfaces.h >>>>>> @@ -329,6 +329,11 @@ typedef struct switch_mm_s { >>>>>> switch_video_profile_t vprofile; >>>>>> switch_video_encode_speed_t vencspd; >>>>>> uint8_t try_hardware_encoder; >>>>>> + int scale_w; >>>>>> + int scale_h; >>>>>> + switch_img_fmt_t fmt; >>>>>> + char *auth_username; >>>>>> + char *auth_password; >>>>>> } switch_mm_t; >>>>>> >>>>>> /*! an abstract representation of a file handle (some parameters >>>>>> based on compat with libsndfile) */ >>>>>> diff --git a/src/mod/applications/mod_av/avformat.c >>>>>> b/src/mod/applications/mod_av/avformat.c >>>>>> index b944625..4b92801 100644 >>>>>> --- a/src/mod/applications/mod_av/avformat.c >>>>>> +++ b/src/mod/applications/mod_av/avformat.c >>>>>> @@ -906,7 +906,7 @@ SWITCH_STANDARD_APP(record_av_function) >>>>>> char codec_str[256]; >>>>>> const AVCodecDescriptor *desc; >>>>>> >>>>>> - if (!strncmp(data, "rtmp://", 7)) { >>>>>> + if (!strncmp(data, "rtmp://", 7) || !strncmp(data, >>>>>> "rtsp://", 7)) { >>>>>> fmt->video_codec = AV_CODEC_ID_H264; >>>>>> fmt->audio_codec = AV_CODEC_ID_AAC; >>>>>> } >>>>>> @@ -1694,9 +1694,20 @@ static switch_status_t >>>>>> av_file_open(switch_file_handle_t *handle, const char *pa >>>>>> return SWITCH_STATUS_GENERR; >>>>>> } else if (handle->stream_name && (!strcasecmp(handle->stream_name, >>>>>> "rtmp") || !strcasecmp(handle->stream_name, "youtube"))) { >>>>>> format = "flv"; >>>>>> - switch_snprintf(file, sizeof(file), "rtmp://%s", >>>>>> path); >>>>>> + >>>>>> + // meh really silly format for the user / pass >>>>>> libav..... >>>>>> + if (handle->mm.auth_username && >>>>>> handle->mm.auth_password) { >>>>>> + switch_snprintf(file, sizeof(file), " >>>>>> rtmp://%s pubUser=%s pubPasswd=%s flashver=FMLE/3.0", path, >>>>>> handle->mm.auth_username, handle->mm.auth_password); >>>>>> + } else { >>>>>> + switch_snprintf(file, sizeof(file), " >>>>>> rtmp://%s", path); >>>>>> + } >>>>>> + >>>>>> + } else if (handle->stream_name && !strcasecmp(handle->stream_name, >>>>>> "rtsp")) { >>>>>> + format = "rtsp"; >>>>>> + switch_snprintf(file, sizeof(file), "rtsp://%s", >>>>>> path); >>>>>> } >>>>>> >>>>>> + >>>>>> ext++; >>>>>> >>>>>> if ((context = (av_file_context_t *)switch_core_alloc(handle->memory_pool, >>>>>> sizeof(av_file_context_t))) == 0) { >>>>>> @@ -1783,7 +1794,7 @@ static switch_status_t av_file_open(switch_file_handle_t >>>>>> *handle, const char *pa >>>>>> if (fmt->video_codec != AV_CODEC_ID_NONE) { >>>>>> const AVCodecDescriptor *desc; >>>>>> >>>>>> - if ((handle->stream_name && >>>>>> (!strcasecmp(handle->stream_name, "rtmp") || >>>>>> !strcasecmp(handle->stream_name, "youtube")))) { >>>>>> + if ((handle->stream_name && >>>>>> (!strcasecmp(handle->stream_name, "rtmp") || >>>>>> !strcasecmp(handle->stream_name, "rtsp") || >>>>>> !strcasecmp(handle->stream_name, "youtube")))) { >>>>>> >>>>>> if (fmt->video_codec != AV_CODEC_ID_H264 ) { >>>>>> fmt->video_codec = AV_CODEC_ID_H264; >>>>>> // force H264 >>>>>> @@ -2525,6 +2536,7 @@ SWITCH_MODULE_LOAD_FUNCTION(mod_avformat_load) >>>>>> >>>>>> supported_formats[i++] = "av"; >>>>>> supported_formats[i++] = "rtmp"; >>>>>> + supported_formats[i++] = "rtsp"; >>>>>> supported_formats[i++] = "mp4"; >>>>>> supported_formats[i++] = "m4a"; >>>>>> supported_formats[i++] = "mov"; >>>>>> diff --git a/src/mod/applications/mod_av/mod_av.c >>>>>> b/src/mod/applications/mod_av/mod_av.c >>>>>> index 3d3bd82..141fcdc 100644 >>>>>> --- a/src/mod/applications/mod_av/mod_av.c >>>>>> +++ b/src/mod/applications/mod_av/mod_av.c >>>>>> @@ -93,7 +93,7 @@ int mod_av_lockmgr_cb(void **m, enum AVLockOp op) >>>>>> static void log_callback(void *ptr, int level, const char *fmt, >>>>>> va_list vl) >>>>>> { >>>>>> switch_log_level_t switch_level = SWITCH_LOG_DEBUG; >>>>>> - >>>>>> + return; >>>>>> /* naggy messages */ >>>>>> if (level == AV_LOG_DEBUG || level == AV_LOG_WARNING) return; >>>>>> >>>>>> diff --git a/src/switch_core_file.c b/src/switch_core_file.c >>>>>> index 46ee539..aff9442 100644 >>>>>> --- a/src/switch_core_file.c >>>>>> +++ b/src/switch_core_file.c >>>>>> @@ -177,6 +177,14 @@ SWITCH_DECLARE(switch_status_t) >>>>>> switch_core_perform_file_open(const char *file, >>>>>> fh->mm.try_hardware_encoder = >>>>>> switch_true(val); >>>>>> } >>>>>> >>>>>> + if ((val = switch_event_get_header(fh->params, >>>>>> "auth_username"))) { >>>>>> + fh->mm.auth_username = >>>>>> switch_core_strdup(fh->memory_pool, val); >>>>>> + } >>>>>> + >>>>>> + if ((val = switch_event_get_header(fh->params, >>>>>> "auth_password"))) { >>>>>> + fh->mm.auth_password = >>>>>> switch_core_strdup(fh->memory_pool, val); >>>>>> + } >>>>>> + >>>>>> if ((val = switch_event_get_header(fh->params, >>>>>> "fps"))) { >>>>>> float ftmp = atof(val); >>>>>> if (ftmp > 0.0f) { >>>>>> (END) >>>>>> >>>>>> >>>>>> I believe that it is the changes in switch_core_file.c that break my >>>>>> utility program, but I can't be sure. Just o add some detail, I have a >>>>>> Digium TCE400 card in my system which handles the G729 conversion. >>>>>> >>>>>> I have no idea of what those auth_username and auth_password >>>>>> parameters are, but that seems to break things for me. Any idea of what I >>>>>> can do to get things working again? >>>>>> >>>>>> I have created a PasteBin with the code of my fs_encode derived >>>>>> utility: https://pastebin.freeswitch.org/view/a3d64c69 It is pretty >>>>>> short. >>>>>> >>>>>> >>>>>> >>>>>> ____________________________________________________________ >>>>>> _____________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at 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 at 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 ♬ @anthmfs ♬ @FreeSWITCH ♬ >>>>> >>>>> ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ >>>>> http://twitter.com/FreeSWITCH >>>>> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ >>>>> * >>>>> >>>>> ClueCon Weekly Development Call >>>>> ☎ sip:888 at conference.freeswitch.org ☎ +19193869900 <(919)%20386-9900> >>>>> >>>>> >>>>> https://www.youtube.com/watch?v=9XXgW34t40s >>>>> https://www.youtube.com/watch?v=NLaDpGQuZDA >>>>> >>>>> ____________________________________________________________ >>>>> _____________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at 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 at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/ >>>>> options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> >>>> -- >>>> Guillermo Ruiz Camauer >>>> >>>> ____________________________________________________________ >>>> _____________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at 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 at 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 ♬ @anthmfs ♬ @FreeSWITCH ♬ >>> >>> ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ >>> http://twitter.com/FreeSWITCH >>> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ >>> * >>> >>> ClueCon Weekly Development Call >>> ☎ sip:888 at conference.freeswitch.org ☎ +19193869900 >>> >>> https://www.youtube.com/watch?v=9XXgW34t40s >>> https://www.youtube.com/watch?v=NLaDpGQuZDA >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at 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 at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Guillermo Ruiz Camauer >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at 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 ♬ @anthmfs ♬ @FreeSWITCH ♬ > > ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ > http://twitter.com/FreeSWITCH > ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ > * > > ClueCon Weekly Development Call > ☎ sip:888 at conference.freeswitch.org ☎ +19193869900 > > https://www.youtube.com/watch?v=9XXgW34t40s > https://www.youtube.com/watch?v=NLaDpGQuZDA > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Guillermo Ruiz Camauer -------------- next part -------------- An HTML attachment was scrubbed... URL: From anthony.minessale at gmail.com Wed Jun 28 02:00:52 2017 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 27 Jun 2017 21:00:52 -0500 Subject: [Freeswitch-users] Problem with fs_encode In-Reply-To: References: Message-ID: On Tue, Jun 27, 2017 at 4:37 PM, Guillermo Ruiz Camauer wrote: > Anthony, > > I wasn't recompiling the app, which I now managed to do and everything is > back to normal. Thanks! > > So if I understand correctly, I must have a version of the app for my > "old" FreeSwitch boxes and another one for my "up to date" FreeSwitch > boxes. Correct? > > I guess it's time to update all boxes... > > Thanks again, > > Guillermo > Yes you need to keep the app up to date as you do with all the other parts of FS such as fs_cli or freeswitch itself. > > On Sat, Jun 24, 2017 at 12:57 AM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> Are you recompiling the app too against the new code. The patch changes >> abi of the fs libs. >> >> >> On Fri, Jun 23, 2017 at 6:35 PM Guillermo Ruiz Camauer < >> grcamauer at gmail.com> wrote: >> >>> Anthony, >>> >>> >>> Removing these lines: >>> >>> + int scale_w; >>> + int scale_h; >>> + switch_img_fmt_t fmt; >>> + char *auth_username; >>> + char *auth_password; >>> >>> from switch_module_interfaces.h >>> >>> makes my program work again. Any idea of what they could be affecting? >>> >>> Guillermo >>> >>> On Fri, Jun 23, 2017 at 8:16 PM, Anthony Minessale < >>> anthony.minessale at gmail.com> wrote: >>> >>>> My advise is to verify by removing bits or even all of the patch and >>>> double check you did not mess up on the bisect. >>>> like if you can manually reproduce going to the version before this >>>> patch and having it work then go to the version with the patch and try >>>> again etc. >>>> >>>> >>>> >>>> On Fri, Jun 23, 2017 at 6:14 PM, Guillermo Ruiz Camauer < >>>> grcamauer at gmail.com> wrote: >>>> >>>>> I will try removing just those lines in switch_core_file.c. Maybe >>>>> it's one of the other changes, I was just guessing... >>>>> >>>>> Guillermo >>>>> >>>>> On Fri, Jun 23, 2017 at 8:03 PM, Anthony Minessale < >>>>> anthony.minessale at gmail.com> wrote: >>>>> >>>>>> Also does removing just that change in switch_core_file.c from latest >>>>>> version actually make it work? It seems unlikely the next line in the file >>>>>> looking for FPS field is no different really. >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> On Fri, Jun 23, 2017 at 6:01 PM, Michael Jerris >>>>>> wrote: >>>>>> >>>>>>> can you drop a core file and see where its stuck? >>>>>>> >>>>>>> On Jun 23, 2017, at 6:57 PM, Guillermo Ruiz Camauer < >>>>>>> grcamauer at gmail.com> wrote: >>>>>>> >>>>>>> I have created a small utility program based on fs_encode which will >>>>>>> take ALL files in a directory and convert them from .WAV to .G729 and .PCMU >>>>>>> formats. The utility is called wavBatchEncode and takes just a directory >>>>>>> path as an argument. >>>>>>> >>>>>>> This program was working very well until I upgraded FreeSwitch. Now >>>>>>> the program seems to hang when it tries to write out the first converted >>>>>>> files. I have to KILL the program from another terminal. It leaves 2 0 >>>>>>> byte files with .PCMU and .G729 extension. >>>>>>> >>>>>>> I have run through a GIT bisect to find where things broke. The >>>>>>> last working version is Version 1.6.16 git ae1cdce 2017-04-11. >>>>>>> The first broken version is 38621e47bad3b63f03a0a27f6ca >>>>>>> 9ed92f6969032. >>>>>>> >>>>>>> I then used GIT DIFF to see if I could see what had happened. There >>>>>>> are very few modifications between these two commits: >>>>>>> >>>>>>> root at fs3:/usr/src/freeswitch.git# git diff ae1cdce >>>>>>> 38621e47bad3b63f03a0a27f6ca9ed92f6969032 >>>>>>> diff --git a/src/include/switch_module_interfaces.h >>>>>>> b/src/include/switch_module_interfaces.h >>>>>>> index e0a5c20..7ca027d 100644 >>>>>>> --- a/src/include/switch_module_interfaces.h >>>>>>> +++ b/src/include/switch_module_interfaces.h >>>>>>> @@ -329,6 +329,11 @@ typedef struct switch_mm_s { >>>>>>> switch_video_profile_t vprofile; >>>>>>> switch_video_encode_speed_t vencspd; >>>>>>> uint8_t try_hardware_encoder; >>>>>>> + int scale_w; >>>>>>> + int scale_h; >>>>>>> + switch_img_fmt_t fmt; >>>>>>> + char *auth_username; >>>>>>> + char *auth_password; >>>>>>> } switch_mm_t; >>>>>>> >>>>>>> /*! an abstract representation of a file handle (some parameters >>>>>>> based on compat with libsndfile) */ >>>>>>> diff --git a/src/mod/applications/mod_av/avformat.c >>>>>>> b/src/mod/applications/mod_av/avformat.c >>>>>>> index b944625..4b92801 100644 >>>>>>> --- a/src/mod/applications/mod_av/avformat.c >>>>>>> +++ b/src/mod/applications/mod_av/avformat.c >>>>>>> @@ -906,7 +906,7 @@ SWITCH_STANDARD_APP(record_av_function) >>>>>>> char codec_str[256]; >>>>>>> const AVCodecDescriptor *desc; >>>>>>> >>>>>>> - if (!strncmp(data, "rtmp://", 7)) { >>>>>>> + if (!strncmp(data, "rtmp://", 7) || !strncmp(data, >>>>>>> "rtsp://", 7)) { >>>>>>> fmt->video_codec = AV_CODEC_ID_H264; >>>>>>> fmt->audio_codec = AV_CODEC_ID_AAC; >>>>>>> } >>>>>>> @@ -1694,9 +1694,20 @@ static switch_status_t >>>>>>> av_file_open(switch_file_handle_t *handle, const char *pa >>>>>>> return SWITCH_STATUS_GENERR; >>>>>>> } else if (handle->stream_name && >>>>>>> (!strcasecmp(handle->stream_name, "rtmp") || >>>>>>> !strcasecmp(handle->stream_name, "youtube"))) { >>>>>>> format = "flv"; >>>>>>> - switch_snprintf(file, sizeof(file), "rtmp://%s", >>>>>>> path); >>>>>>> + >>>>>>> + // meh really silly format for the user / pass >>>>>>> libav..... >>>>>>> + if (handle->mm.auth_username && >>>>>>> handle->mm.auth_password) { >>>>>>> + switch_snprintf(file, sizeof(file), " >>>>>>> rtmp://%s pubUser=%s pubPasswd=%s flashver=FMLE/3.0", path, >>>>>>> handle->mm.auth_username, handle->mm.auth_password); >>>>>>> + } else { >>>>>>> + switch_snprintf(file, sizeof(file), " >>>>>>> rtmp://%s", path); >>>>>>> + } >>>>>>> + >>>>>>> + } else if (handle->stream_name && >>>>>>> !strcasecmp(handle->stream_name, "rtsp")) { >>>>>>> + format = "rtsp"; >>>>>>> + switch_snprintf(file, sizeof(file), "rtsp://%s", >>>>>>> path); >>>>>>> } >>>>>>> >>>>>>> + >>>>>>> ext++; >>>>>>> >>>>>>> if ((context = (av_file_context_t >>>>>>> *)switch_core_alloc(handle->memory_pool, >>>>>>> sizeof(av_file_context_t))) == 0) { >>>>>>> @@ -1783,7 +1794,7 @@ static switch_status_t >>>>>>> av_file_open(switch_file_handle_t *handle, const char *pa >>>>>>> if (fmt->video_codec != AV_CODEC_ID_NONE) { >>>>>>> const AVCodecDescriptor *desc; >>>>>>> >>>>>>> - if ((handle->stream_name && >>>>>>> (!strcasecmp(handle->stream_name, "rtmp") || >>>>>>> !strcasecmp(handle->stream_name, "youtube")))) { >>>>>>> + if ((handle->stream_name && >>>>>>> (!strcasecmp(handle->stream_name, "rtmp") || >>>>>>> !strcasecmp(handle->stream_name, "rtsp") || >>>>>>> !strcasecmp(handle->stream_name, "youtube")))) { >>>>>>> >>>>>>> if (fmt->video_codec != AV_CODEC_ID_H264 ) { >>>>>>> fmt->video_codec = AV_CODEC_ID_H264; >>>>>>> // force H264 >>>>>>> @@ -2525,6 +2536,7 @@ SWITCH_MODULE_LOAD_FUNCTION(mod_avformat_load) >>>>>>> >>>>>>> supported_formats[i++] = "av"; >>>>>>> supported_formats[i++] = "rtmp"; >>>>>>> + supported_formats[i++] = "rtsp"; >>>>>>> supported_formats[i++] = "mp4"; >>>>>>> supported_formats[i++] = "m4a"; >>>>>>> supported_formats[i++] = "mov"; >>>>>>> diff --git a/src/mod/applications/mod_av/mod_av.c >>>>>>> b/src/mod/applications/mod_av/mod_av.c >>>>>>> index 3d3bd82..141fcdc 100644 >>>>>>> --- a/src/mod/applications/mod_av/mod_av.c >>>>>>> +++ b/src/mod/applications/mod_av/mod_av.c >>>>>>> @@ -93,7 +93,7 @@ int mod_av_lockmgr_cb(void **m, enum AVLockOp op) >>>>>>> static void log_callback(void *ptr, int level, const char *fmt, >>>>>>> va_list vl) >>>>>>> { >>>>>>> switch_log_level_t switch_level = SWITCH_LOG_DEBUG; >>>>>>> - >>>>>>> + return; >>>>>>> /* naggy messages */ >>>>>>> if (level == AV_LOG_DEBUG || level == AV_LOG_WARNING) return; >>>>>>> >>>>>>> diff --git a/src/switch_core_file.c b/src/switch_core_file.c >>>>>>> index 46ee539..aff9442 100644 >>>>>>> --- a/src/switch_core_file.c >>>>>>> +++ b/src/switch_core_file.c >>>>>>> @@ -177,6 +177,14 @@ SWITCH_DECLARE(switch_status_t) >>>>>>> switch_core_perform_file_open(const char *file, >>>>>>> fh->mm.try_hardware_encoder = >>>>>>> switch_true(val); >>>>>>> } >>>>>>> >>>>>>> + if ((val = switch_event_get_header(fh->params, >>>>>>> "auth_username"))) { >>>>>>> + fh->mm.auth_username = >>>>>>> switch_core_strdup(fh->memory_pool, val); >>>>>>> + } >>>>>>> + >>>>>>> + if ((val = switch_event_get_header(fh->params, >>>>>>> "auth_password"))) { >>>>>>> + fh->mm.auth_password = >>>>>>> switch_core_strdup(fh->memory_pool, val); >>>>>>> + } >>>>>>> + >>>>>>> if ((val = switch_event_get_header(fh->params, >>>>>>> "fps"))) { >>>>>>> float ftmp = atof(val); >>>>>>> if (ftmp > 0.0f) { >>>>>>> (END) >>>>>>> >>>>>>> >>>>>>> I believe that it is the changes in switch_core_file.c that break my >>>>>>> utility program, but I can't be sure. Just o add some detail, I have a >>>>>>> Digium TCE400 card in my system which handles the G729 conversion. >>>>>>> >>>>>>> I have no idea of what those auth_username and auth_password >>>>>>> parameters are, but that seems to break things for me. Any idea of what I >>>>>>> can do to get things working again? >>>>>>> >>>>>>> I have created a PasteBin with the code of my fs_encode derived >>>>>>> utility: https://pastebin.freeswitch.org/view/a3d64c69 It is >>>>>>> pretty short. >>>>>>> >>>>>>> >>>>>>> >>>>>>> ____________________________________________________________ >>>>>>> _____________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at 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 at 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 ♬ @anthmfs ♬ @FreeSWITCH ♬ >>>>>> >>>>>> ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ >>>>>> http://twitter.com/FreeSWITCH >>>>>> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ >>>>>> * >>>>>> >>>>>> ClueCon Weekly Development Call >>>>>> ☎ sip:888 at conference.freeswitch.org ☎ +19193869900 >>>>>> <(919)%20386-9900> >>>>>> >>>>>> https://www.youtube.com/watch?v=9XXgW34t40s >>>>>> https://www.youtube.com/watch?v=NLaDpGQuZDA >>>>>> >>>>>> ____________________________________________________________ >>>>>> _____________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at 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 at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ >>>>>> freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>> >>>>> >>>>> >>>>> -- >>>>> Guillermo Ruiz Camauer >>>>> >>>>> ____________________________________________________________ >>>>> _____________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at 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 at 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 ♬ @anthmfs ♬ @FreeSWITCH ♬ >>>> >>>> ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ >>>> http://twitter.com/FreeSWITCH >>>> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ >>>> * >>>> >>>> ClueCon Weekly Development Call >>>> ☎ sip:888 at conference.freeswitch.org ☎ +19193869900 <(919)%20386-9900> >>>> >>>> https://www.youtube.com/watch?v=9XXgW34t40s >>>> https://www.youtube.com/watch?v=NLaDpGQuZDA >>>> >>>> ____________________________________________________________ >>>> _____________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at 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 at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/ >>>> freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> >>> -- >>> Guillermo Ruiz Camauer >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at 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 at 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 ♬ @anthmfs ♬ @FreeSWITCH ♬ >> >> ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ >> http://twitter.com/FreeSWITCH >> ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ >> * >> >> ClueCon Weekly Development Call >> ☎ sip:888 at conference.freeswitch.org ☎ +19193869900 <(919)%20386-9900> >> >> https://www.youtube.com/watch?v=9XXgW34t40s >> https://www.youtube.com/watch?v=NLaDpGQuZDA >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Guillermo Ruiz Camauer > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at 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 ♬ @anthmfs ♬ @FreeSWITCH ♬ ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ * ClueCon Weekly Development Call ☎ sip:888 at conference.freeswitch.org ☎ +19193869900 https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA -------------- next part -------------- An HTML attachment was scrubbed... URL: From thetsinling at outlook.com Wed Jun 28 04:25:23 2017 From: thetsinling at outlook.com (bob. chen) Date: Wed, 28 Jun 2017 04:25:23 +0000 Subject: [Freeswitch-users] VOIP-GSM Gateway configuration Error In-Reply-To: References: Message-ID: https://freeswitch.org/confluence/display/FREESWITCH/Dinstar+GSM+gateway+FreeSwitch+HowTo try peer to peer ;) From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Deepika Yadav Sent: Tuesday, June 27, 2017 7:06 PM To: FreeSWITCH Users Help Subject: [Freeswitch-users] VOIP-GSM Gateway configuration Error Hi, I have been trying for a while the configuration of a VOIP-GSM gateway that I have bought but the registration fails every time. Error at Freeswitch console 2017-06-27 16:27:10.191913 [WARNING] sofia_reg.c:2906 Can't find user [testadmin at 192.168.2.71] from 192.168.2.69 You must define a domain called '192.168.2.71' in your directory and add a user with the id="testadmin" attribute and you must configure your device to use the proper domain in it's authentication credentials. 2017-06-27 16:27:14.071916 [ERR] sofia_reg.c:2447 MySIP Failed Registration with status Request Timeout [408]. failure #1 2017-06-27 16:27:14.111913 [WARNING] sofia_reg.c:505 MySIP Failed Registration [408], setting retry to 30 seconds. Here '192.168.2.71' is the local IP of the Freeswitch server and 192.168.2.69 that of the gateway attached to the LAN port of the Freeswitch server. I have put SIP UserID in the gateway admin panel as "testadmin" alongwith its passoword. my external SIP profile is as follows: I have tried number of permutations and combinations but the gateway is not getting registered. Regards Deepika https://www.iiitd.edu.in/~deepikay/ -------------- next part -------------- An HTML attachment was scrubbed... URL: From bipin at xbipin.com Wed Jun 28 08:23:41 2017 From: bipin at xbipin.com (Bipin Patel) Date: Wed, 28 Jun 2017 12:23:41 +0400 Subject: [Freeswitch-users] keep limit counter active across dialplans when used in attended transfers In-Reply-To: References: <15ce0d1aa40.279b.b07ebdf329620b8089087c7205b03f01@xbipin.com> <752b7178-f635-a544-05a6-df985bfa851f@xbipin.com> <6418EDCA-2820-4C7B-B263-39A701FCF409@vallimamod.org> Message-ID: <10224122-2181-5235-0a16-4bcfdf6df46b@xbipin.com> hi, i got the thing to work, it works for most parts only in one case it doesnt, i insert a hash key/value before the bridge and in the bridge i use api_hangup_hook=hash delete/max_calls/${destination_number} but suppose if the person being called isnt registered and the bridge fails then api_hangup_hook never executes. Is there any other api like fail hook or execute on fail that i can use so i can delete the hash ke/value? Regards, Bipin ------------------------------------------------------------------------ -------- Original Message -------- Subject: Re: [Freeswitch-users] keep limit counter active across dialplans when used in attended transfers From: Bipin Patel To: FreeSWITCH Users Help Date: 6/27/2017, 11:10:47 AM > hi, > > and also hangup_after_bridge doesnt work to delete the key, need to > use set_zombie_exec > > > Regards, > Bipin > > > ------------------------------------------------------------------------ > -------- Original Message -------- > Subject: Re: [Freeswitch-users] keep limit counter active across > dialplans when used in attended transfers > From: Bipin Patel > To: FreeSWITCH Users Help > Date: 6/27/2017, 11:06:03 AM >> hi, >> >> yes i was trying that but the problem occurs when ur doing attended >> transfer because there when we set the hash key to oncall in the >> features dialplan, the bridge is actually a att_xfer so it doesnt run >> the delete command after C is hungup >> >> >> Regards, >> Bipin >> >> >> ------------------------------------------------------------------------ >> -------- Original Message -------- >> Subject: Re: [Freeswitch-users] keep limit counter active across >> dialplans when used in attended transfers >> From: Vallimamod Abdullah >> To: FreeSWITCH Users Help >> Date: 6/27/2017, 10:58:13 AM >>> Hi, >>> >>> You can set hangup_after_bridge to false and delete the key after >>> the bridge command. >>> >>> Best Regards, >>> -- >>> Vallimamod Abdullah >>> SIP Solutions >>> Consulting VOIP >>> vma at sipsolutions.fr >>> . >>> >>>> On 26 Jun 2017, at 15:59, Bipin Patel >>> > wrote: >>>> >>>> hi, >>>> >>>> i was testing mod_hash but how do we delete the key when the caller >>>> has hung up causing originator cancel because the dialplan doesnt >>>> execute any further, it does when the call is answered then hungup. >>>> >>>> Im assuming mod_hash works across dialplans because i didnt find >>>> anything in docs that would suggest it wont >>>> >>>> >>>> Regards, >>>> Bipin >>>> >>>> >>>> ------------------------------------------------------------------------ >>>> -------- Original Message -------- >>>> Subject: Re: [Freeswitch-users] keep limit counter active across >>>> dialplans when used in attended transfers >>>> From: Bipin Patel >>>> To: FreeSWITCH Users Help >>>> Date: 6/25/2017, 11:54:16 PM >>>>> >>>>> Hi, >>>>> >>>>> Well I could use it but with limit the counter decrementing etc is >>>>> automatic so it's much simpler and reduces the dialplan required. >>>>> >>>>> In the dialplan I provided how do you think I could use it >>>>> directly without adding any more extensions in the two dialplan, >>>>> any help would be appreciated. What I need to do is increment >>>>> counter for caller and callee as well as callee transferred to and >>>>> check that counter before bridging so change ring back can be called. >>>>> >>>>> On June 25, 2017 8:43:29 PM Vallimamod >>>>> Abdullahwrote: >>>>> >>>>>> Hi, >>>>>> >>>>>> Is there any reason why you don't use the hash application >>>>>> directly instead of limit? >>>>>> IMHO it is more suited to your use case. >>>>>> >>>>>> https://freeswitch.org/confluence/display/FREESWITCH/mod_hash >>>>>> >>>>>> Best Regards, >>>>>> -- >>>>>> Vallimamod Abdullah >>>>>> SIP Solutions >>>>>> vma at sipsolutions.fr >>>>>> . >>>>>> >>>>>> >>>>>>> On 25 Jun 2017, at 17:10, Bipin Patel >>>>>> > wrote: >>>>>>> >>>>>>> hi, >>>>>>> >>>>>>> i have a requirement where i need to generate a different >>>>>>> ringtone if that ext is on another call and i do it using limit >>>>>>> by setting a counter for the caller and callee and then checking >>>>>>> the counter if it has a value of 1 which would mean that ext is >>>>>>> on anohter call so generate a different ringtone and this work fine. >>>>>>> >>>>>>> The problem being if A calls B and B attended transfers the call >>>>>>> to C then the counter for C is incremented in the features >>>>>>> dialplan but it doesnt reflect in the internal dialplan and i >>>>>>> read that active limits dont remain global across dialplans so >>>>>>> is there anyway to keep the limit counter active. i tried >>>>>>> limit_ignore_transfer but that doesnt seem to work, im on the FS >>>>>>> master >>>>>>> >>>>>>> Below is my internal and features dialplan, any pointers would >>>>>>> be helpful >>>>>>> >>>>>>> >>>>>>> INTERNAL DIALPLAN >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> expression="^ChangeRingBack$"/> >>>>>>> >>>>>>> >>>>>> data="transfer_ringback=%(250,250,400);%(250,3500,400)"/> >>>>>>> >>>>>> data="ringback=%(250,250,400);%(250,3500,400)"/> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> data="transfer_ringback=%(400,200,400,375);%(400,2000,400,375)"/> >>>>>>> >>>>>>> >>>>>> data="ringback=%(400,200,400,375);%(400,2000,400,375)"/> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> FEATURES DIALPLAN >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> data="transfer_ringback=%(400,200,400,375);%(400,2000,400,375)"/> >>>>>>> >>>>>>> >>>>>> data="ringback=%(400,200,400,375);%(400,2000,400,375)"/> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>> data="sofia/internal/${digits}%$${domain}"/> >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> -- >>>>>>> Regards, >>>>>>> Bipin >>>>>>> >>>>>>> >>>>>>> ------------------------------------------------------------------------ >>>>>>> _________________________________________________________________________ >>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>> consulting at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From rick at magicmail.mooo.com Wed Jun 28 11:43:36 2017 From: rick at magicmail.mooo.com (Rick Jarvis) Date: Wed, 28 Jun 2017 12:43:36 +0100 Subject: [Freeswitch-users] FS service issues (systemd) Message-ID: <6B172391-E75C-4CA2-A2DE-2C0A652C19EC@magicmail.mooo.com> Got a weird issue where FS isn’t starting up correctly. Looking at the logs, it appears just to be attempting to restart SIP profiles (and failing), and that’s about it. There’s no socket for fs_cli, and a simple ‘systemctl restart’ fixes it. Anyone had something like this? I’ve tried fixing it with systemd after / requires, but it hasn’t helped. Here’s my .service file (not sure if this is the issue, I'd welcome any comments / better versions?) [Unit] Description=freeswitch After=syslog.target network-online.target local-fs.target [Service] ; service Type=forking PIDFile=/run/freeswitch/freeswitch.pid PermissionsStartOnly=true ExecStart=/usr/bin/freeswitch -ncwait -nonat ExecStop=/usr/bin/fs_cli -x shutdown TimeoutSec=45s Restart=always ; exec RuntimeDirectory=freeswitch RuntimeDirectoryMode=0755 User=freeswitch Group=freeswitch LimitCORE=infinity LimitNOFILE=100000 LimitNPROC=60000 ;LimitSTACK=240 LimitRTPRIO=infinity LimitRTTIME=7000000 IOSchedulingClass=realtime IOSchedulingPriority=2 CPUSchedulingPolicy=rr CPUSchedulingPriority=89 UMask=0007 [Install] -------------- next part -------------- An HTML attachment was scrubbed... URL: From rmundkowsky at ets.org Wed Jun 28 13:18:27 2017 From: rmundkowsky at ets.org (Mundkowsky, Robert) Date: Wed, 28 Jun 2017 13:18:27 +0000 Subject: [Freeswitch-users] FS service issues (systemd) In-Reply-To: <6B172391-E75C-4CA2-A2DE-2C0A652C19EC@magicmail.mooo.com> References: <6B172391-E75C-4CA2-A2DE-2C0A652C19EC@magicmail.mooo.com> Message-ID: Yes. I think we had a similar problem in ,I think, two situations. 1) The file system database can get messed up. a. Stop FS b. You can clear then via rm /export/Apps/freeswitch/var/lib/freeswitch/db/* c. Restart FS 2) If there is an error in the configuration files. a. Take a look at /export/Apps/freeswitch/var/log/freeswitch/freeswitch.xml.fsxml to see if there are any errors in XML. You can get more details of start problem via: journalctl -xn systemctl status freeswitch.service if you do not get enough details, try starting FS directly and you might get more details via: /…/bin/freeswitch Robert Mundkowsky From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rick Jarvis Sent: Wednesday, June 28, 2017 7:44 AM To: FreeSWITCH Users Help Subject: [Freeswitch-users] FS service issues (systemd) Got a weird issue where FS isn’t starting up correctly. Looking at the logs, it appears just to be attempting to restart SIP profiles (and failing), and that’s about it. There’s no socket for fs_cli, and a simple ‘systemctl restart’ fixes it. Anyone had something like this? I’ve tried fixing it with systemd after / requires, but it hasn’t helped. Here’s my .service file (not sure if this is the issue, I'd welcome any comments / better versions?) [Unit] Description=freeswitch After=syslog.target network-online.target local-fs.target [Service] ; service Type=forking PIDFile=/run/freeswitch/freeswitch.pid PermissionsStartOnly=true ExecStart=/usr/bin/freeswitch -ncwait -nonat ExecStop=/usr/bin/fs_cli -x shutdown TimeoutSec=45s Restart=always ; exec RuntimeDirectory=freeswitch RuntimeDirectoryMode=0755 User=freeswitch Group=freeswitch LimitCORE=infinity LimitNOFILE=100000 LimitNPROC=60000 ;LimitSTACK=240 LimitRTPRIO=infinity LimitRTTIME=7000000 IOSchedulingClass=realtime IOSchedulingPriority=2 CPUSchedulingPolicy=rr CPUSchedulingPriority=89 UMask=0007 [Install] ________________________________ This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited. Thank you for your compliance. ________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL: From hunterj91 at hotmail.com Wed Jun 28 13:33:00 2017 From: hunterj91 at hotmail.com (Jonathan Hunter) Date: Wed, 28 Jun 2017 13:33:00 +0000 Subject: [Freeswitch-users] Is it possible to send an SIP UPDATE message via sendevent ? Message-ID: Hi Guys, I have a quick question, I can see from documentation you can send SIP messages via sendevent and the ESL. Is it possible to send a SIP UPDATE message with this method and if so are there any configuration options I need to be aware of? Essentially we want to send an UPDATE after answer of a call to flag a change. Thanks Jon -------------- next part -------------- An HTML attachment was scrubbed... URL: From rick at magicmail.mooo.com Wed Jun 28 13:42:41 2017 From: rick at magicmail.mooo.com (Rick Jarvis) Date: Wed, 28 Jun 2017 14:42:41 +0100 Subject: [Freeswitch-users] FS service issues (systemd) In-Reply-To: References: <6B172391-E75C-4CA2-A2DE-2C0A652C19EC@magicmail.mooo.com> Message-ID: <518D9916-4387-4C50-9932-DE812A12856B@magicmail.mooo.com> I’ve tried deleting the db files but no difference unfortunately. Starting manually or even just restarting the service works, so I believe the XML is all as it should be. This is my output from systemctl when it’s in this ‘hung' state: ● freeswitch.service - freeswitch Loaded: loaded (/etc/systemd/system/freeswitch.service; enabled) Active: active (running) since Wed 2017-06-28 14:36:23 BST; 3min 27s ago Process: 452 ExecStart=/usr/bin/freeswitch -ncwait -nonat (code=exited, status=0/SUCCESS) Main PID: 472 (freeswitch) CGroup: /system.slice/freeswitch.service └─472 /usr/bin/freeswitch -ncwait -nonat Jun 28 14:36:20 server systemd[1]: Starting freeswitch... Jun 28 14:36:20 server freeswitch[452]: 472 Backgrounding. Jun 28 14:36:23 server freeswitch[452]: FreeSWITCH[452] Waiting for background process pid:472 to be ready..... Jun 28 14:36:23 server freeswitch[452]: FreeSWITCH[452] System Ready pid:472 Jun 28 14:36:23 server systemd[1]: Started freeswitch. > On 28 Jun 2017, at 14:18, Mundkowsky, Robert wrote: > > Yes. I think we had a similar problem in ,I think, two situations. <> > > 1) The file system database can get messed up. > a. Stop FS > b. You can clear then via rm /export/Apps/freeswitch/var/lib/freeswitch/db/* > c. Restart FS > 2) If there is an error in the configuration files. > a. Take a look at /export/Apps/freeswitch/var/log/freeswitch/freeswitch.xml.fsxml to see if there are any errors in XML. > > You can get more details of start problem via: > journalctl -xn > systemctl status freeswitch.service > > if you do not get enough details, try starting FS directly and you might get more details via: > > /…/bin/freeswitch > > > Robert Mundkowsky > > From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org ] On Behalf Of Rick Jarvis > Sent: Wednesday, June 28, 2017 7:44 AM > To: FreeSWITCH Users Help > > Subject: [Freeswitch-users] FS service issues (systemd) > > > Got a weird issue where FS isn’t starting up correctly. Looking at the logs, it appears just to be attempting to restart SIP profiles (and failing), and that’s about it. There’s no socket for fs_cli, and a simple ‘systemctl restart’ fixes it. Anyone had something like this? I’ve tried fixing it with systemd after / requires, but it hasn’t helped. > > Here’s my .service file (not sure if this is the issue, I'd welcome any comments / better versions?) > > [Unit] > Description=freeswitch > After=syslog.target network-online.target local-fs.target > > [Service] > ; service > Type=forking > PIDFile=/run/freeswitch/freeswitch.pid > PermissionsStartOnly=true > ExecStart=/usr/bin/freeswitch -ncwait -nonat > ExecStop=/usr/bin/fs_cli -x shutdown > TimeoutSec=45s > Restart=always > ; exec > RuntimeDirectory=freeswitch > RuntimeDirectoryMode=0755 > User=freeswitch > Group=freeswitch > LimitCORE=infinity > LimitNOFILE=100000 > LimitNPROC=60000 > ;LimitSTACK=240 > LimitRTPRIO=infinity > LimitRTTIME=7000000 > IOSchedulingClass=realtime > IOSchedulingPriority=2 > CPUSchedulingPolicy=rr > CPUSchedulingPriority=89 > UMask=0007 > > [Install] > > > > > > > > This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited. > > > Thank you for your compliance. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From ljjimenez at gmail.com Wed Jun 28 13:52:21 2017 From: ljjimenez at gmail.com (Luis Jimenez) Date: Wed, 28 Jun 2017 09:52:21 -0400 Subject: [Freeswitch-users] FS service issues (systemd) In-Reply-To: <518D9916-4387-4C50-9932-DE812A12856B@magicmail.mooo.com> References: <6B172391-E75C-4CA2-A2DE-2C0A652C19EC@magicmail.mooo.com> <518D9916-4387-4C50-9932-DE812A12856B@magicmail.mooo.com> Message-ID: <4773C447-BBDD-46BC-8584-4A776A726FF5@gmail.com> What about the freeswitch.log file when fs is hung? > On Jun 28, 2017, at 09:42, Rick Jarvis wrote: > > I’ve tried deleting the db files but no difference unfortunately. Starting manually or even just restarting the service works, so I believe the XML is all as it should be. This is my output from systemctl when it’s in this ‘hung' state: > > ● freeswitch.service - freeswitch > Loaded: loaded (/etc/systemd/system/freeswitch.service; enabled) > Active: active (running) since Wed 2017-06-28 14:36:23 BST; 3min 27s ago > Process: 452 ExecStart=/usr/bin/freeswitch -ncwait -nonat (code=exited, status=0/SUCCESS) > Main PID: 472 (freeswitch) > CGroup: /system.slice/freeswitch.service > └─472 /usr/bin/freeswitch -ncwait -nonat > > Jun 28 14:36:20 server systemd[1]: Starting freeswitch... > Jun 28 14:36:20 server freeswitch[452]: 472 Backgrounding. > Jun 28 14:36:23 server freeswitch[452]: FreeSWITCH[452] Waiting for background process pid:472 to be ready..... > Jun 28 14:36:23 server freeswitch[452]: FreeSWITCH[452] System Ready pid:472 > Jun 28 14:36:23 server systemd[1]: Started freeswitch. > > >> On 28 Jun 2017, at 14:18, Mundkowsky, Robert wrote: >> >> Yes. I think we had a similar problem in ,I think, two situations. >> >> 1) The file system database can get messed up. >> a. Stop FS >> b. You can clear then via rm /export/Apps/freeswitch/var/lib/freeswitch/db/* >> c. Restart FS >> 2) If there is an error in the configuration files. >> a. Take a look at /export/Apps/freeswitch/var/log/freeswitch/freeswitch.xml.fsxml to see if there are any errors in XML. >> >> You can get more details of start problem via: >> journalctl -xn >> systemctl status freeswitch.service >> >> if you do not get enough details, try starting FS directly and you might get more details via: >> >> /…/bin/freeswitch >> >> >> Robert Mundkowsky >> >> From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rick Jarvis >> Sent: Wednesday, June 28, 2017 7:44 AM >> To: FreeSWITCH Users Help >> Subject: [Freeswitch-users] FS service issues (systemd) >> >> >> Got a weird issue where FS isn’t starting up correctly. Looking at the logs, it appears just to be attempting to restart SIP profiles (and failing), and that’s about it. There’s no socket for fs_cli, and a simple ‘systemctl restart’ fixes it. Anyone had something like this? I’ve tried fixing it with systemd after / requires, but it hasn’t helped. >> >> Here’s my .service file (not sure if this is the issue, I'd welcome any comments / better versions?) >> >> [Unit] >> Description=freeswitch >> After=syslog.target network-online.target local-fs.target >> >> [Service] >> ; service >> Type=forking >> PIDFile=/run/freeswitch/freeswitch.pid >> PermissionsStartOnly=true >> ExecStart=/usr/bin/freeswitch -ncwait -nonat >> ExecStop=/usr/bin/fs_cli -x shutdown >> TimeoutSec=45s >> Restart=always >> ; exec >> RuntimeDirectory=freeswitch >> RuntimeDirectoryMode=0755 >> User=freeswitch >> Group=freeswitch >> LimitCORE=infinity >> LimitNOFILE=100000 >> LimitNPROC=60000 >> ;LimitSTACK=240 >> LimitRTPRIO=infinity >> LimitRTTIME=7000000 >> IOSchedulingClass=realtime >> IOSchedulingPriority=2 >> CPUSchedulingPolicy=rr >> CPUSchedulingPriority=89 >> UMask=0007 >> >> [Install] >> >> >> >> >> >> >> >> This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited. >> >> >> Thank you for your compliance. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From rick at magicmail.mooo.com Wed Jun 28 14:32:38 2017 From: rick at magicmail.mooo.com (Rick Jarvis) Date: Wed, 28 Jun 2017 15:32:38 +0100 Subject: [Freeswitch-users] FS service issues (systemd) In-Reply-To: <4773C447-BBDD-46BC-8584-4A776A726FF5@gmail.com> References: <6B172391-E75C-4CA2-A2DE-2C0A652C19EC@magicmail.mooo.com> <518D9916-4387-4C50-9932-DE812A12856B@magicmail.mooo.com> <4773C447-BBDD-46BC-8584-4A776A726FF5@gmail.com> Message-ID: <828993D4-5964-4EE7-87B6-D15830E045D3@magicmail.mooo.com> Just looping with failed registrations: 2017-06-28 15:29:42.738155 [NOTICE] sofia_reg.c:448 Registering connection1 2017-06-28 15:29:42.738155 [NOTICE] sofia_reg.c:448 Registering connection2 2017-06-28 15:29:42.738155 [NOTICE] sofia_reg.c:448 Registering xprov 2017-06-28 15:29:42.738155 [ERR] sofia_reg.c:2392 connection2 Failed Registration with status Service Unavailable [503]. failure #101 2017-06-28 15:29:42.738155 [ERR] sofia_reg.c:2392 xprov Failed Registration with status Service Unavailable [503]. failure #101 2017-06-28 15:29:42.838117 [ERR] sofia_reg.c:2392 connection1 Failed Registration with status Service Unavailable [503]. failure #101 2017-06-28 15:29:43.738108 [WARNING] sofia_reg.c:505 connection1 Failed Registration [503], setting retry to 30 seconds. 2017-06-28 15:29:43.738108 [WARNING] sofia_reg.c:505 connection2 Failed Registration [503], setting retry to 30 seconds. 2017-06-28 15:29:43.738108 [WARNING] sofia_reg.c:505 xprov Failed Registration [503], setting retry to 30 seconds. > On 28 Jun 2017, at 14:52, Luis Jimenez wrote: > > What about the freeswitch.log file when fs is hung? > > > On Jun 28, 2017, at 09:42, Rick Jarvis > wrote: > >> I’ve tried deleting the db files but no difference unfortunately. Starting manually or even just restarting the service works, so I believe the XML is all as it should be. This is my output from systemctl when it’s in this ‘hung' state: >> >> ● freeswitch.service - freeswitch >> Loaded: loaded (/etc/systemd/system/freeswitch.service; enabled) >> Active: active (running) since Wed 2017-06-28 14:36:23 BST; 3min 27s ago >> Process: 452 ExecStart=/usr/bin/freeswitch -ncwait -nonat (code=exited, status=0/SUCCESS) >> Main PID: 472 (freeswitch) >> CGroup: /system.slice/freeswitch.service >> └─472 /usr/bin/freeswitch -ncwait -nonat >> >> Jun 28 14:36:20 server systemd[1]: Starting freeswitch... >> Jun 28 14:36:20 server freeswitch[452]: 472 Backgrounding. >> Jun 28 14:36:23 server freeswitch[452]: FreeSWITCH[452] Waiting for background process pid:472 to be ready..... >> Jun 28 14:36:23 server freeswitch[452]: FreeSWITCH[452] System Ready pid:472 >> Jun 28 14:36:23 server systemd[1]: Started freeswitch. >> >> >>> On 28 Jun 2017, at 14:18, Mundkowsky, Robert > wrote: >>> >>> Yes. I think we had a similar problem in ,I think, two situations. <> >>> >>> 1) The file system database can get messed up. >>> a. Stop FS >>> b. You can clear then via rm /export/Apps/freeswitch/var/lib/freeswitch/db/* >>> c. Restart FS >>> 2) If there is an error in the configuration files. >>> a. Take a look at /export/Apps/freeswitch/var/log/freeswitch/freeswitch.xml.fsxml to see if there are any errors in XML. >>> >>> You can get more details of start problem via: >>> journalctl -xn >>> systemctl status freeswitch.service >>> >>> if you do not get enough details, try starting FS directly and you might get more details via: >>> >>> /…/bin/freeswitch >>> >>> >>> Robert Mundkowsky >>> >>> From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org ] On Behalf Of Rick Jarvis >>> Sent: Wednesday, June 28, 2017 7:44 AM >>> To: FreeSWITCH Users Help > >>> Subject: [Freeswitch-users] FS service issues (systemd) >>> >>> >>> Got a weird issue where FS isn’t starting up correctly. Looking at the logs, it appears just to be attempting to restart SIP profiles (and failing), and that’s about it. There’s no socket for fs_cli, and a simple ‘systemctl restart’ fixes it. Anyone had something like this? I’ve tried fixing it with systemd after / requires, but it hasn’t helped. >>> >>> Here’s my .service file (not sure if this is the issue, I'd welcome any comments / better versions?) >>> >>> [Unit] >>> Description=freeswitch >>> After=syslog.target network-online.target local-fs.target >>> >>> [Service] >>> ; service >>> Type=forking >>> PIDFile=/run/freeswitch/freeswitch.pid >>> PermissionsStartOnly=true >>> ExecStart=/usr/bin/freeswitch -ncwait -nonat >>> ExecStop=/usr/bin/fs_cli -x shutdown >>> TimeoutSec=45s >>> Restart=always >>> ; exec >>> RuntimeDirectory=freeswitch >>> RuntimeDirectoryMode=0755 >>> User=freeswitch >>> Group=freeswitch >>> LimitCORE=infinity >>> LimitNOFILE=100000 >>> LimitNPROC=60000 >>> ;LimitSTACK=240 >>> LimitRTPRIO=infinity >>> LimitRTTIME=7000000 >>> IOSchedulingClass=realtime >>> IOSchedulingPriority=2 >>> CPUSchedulingPolicy=rr >>> CPUSchedulingPriority=89 >>> UMask=0007 >>> >>> [Install] >>> >>> >>> >>> >>> >>> >>> >>> This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited. >>> >>> >>> Thank you for your compliance. >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at 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 at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From gmaruzz at gmail.com Wed Jun 28 14:44:20 2017 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Wed, 28 Jun 2017 16:44:20 +0200 Subject: [Freeswitch-users] FS service issues (systemd) In-Reply-To: <828993D4-5964-4EE7-87B6-D15830E045D3@magicmail.mooo.com> References: <6B172391-E75C-4CA2-A2DE-2C0A652C19EC@magicmail.mooo.com> <518D9916-4387-4C50-9932-DE812A12856B@magicmail.mooo.com> <4773C447-BBDD-46BC-8584-4A776A726FF5@gmail.com> <828993D4-5964-4EE7-87B6-D15830E045D3@magicmail.mooo.com> Message-ID: What exact platform are you using? Os/distro? FS version? Any virtualization? Has ever started correctly? Have you started it from command line (maybe as root user) and then the problems started to appear? Have you tried to reinstall from scratch following strictly the confluence instructions? Maybe file/directories permission problem? -giovanni On 28 June 2017 at 16:32, Rick Jarvis wrote: > Just looping with failed registrations: > > 2017-06-28 15:29:42.738155 [NOTICE] sofia_reg.c:448 Registering connection1 > 2017-06-28 15:29:42.738155 [NOTICE] sofia_reg.c:448 Registering connection2 > 2017-06-28 15:29:42.738155 [NOTICE] sofia_reg.c:448 Registering xprov > 2017-06-28 15:29:42.738155 [ERR] sofia_reg.c:2392 connection2 Failed > Registration with status Service Unavailable [503]. failure #101 > 2017-06-28 15:29:42.738155 [ERR] sofia_reg.c:2392 xprov Failed > Registration with status Service Unavailable [503]. failure #101 > 2017-06-28 15:29:42.838117 [ERR] sofia_reg.c:2392 connection1 Failed > Registration with status Service Unavailable [503]. failure #101 > 2017-06-28 15:29:43.738108 [WARNING] sofia_reg.c:505 connection1 Failed > Registration [503], setting retry to 30 seconds. > 2017-06-28 15:29:43.738108 [WARNING] sofia_reg.c:505 connection2 Failed > Registration [503], setting retry to 30 seconds. > 2017-06-28 15:29:43.738108 [WARNING] sofia_reg.c:505 xprov Failed > Registration [503], setting retry to 30 seconds. > > On 28 Jun 2017, at 14:52, Luis Jimenez wrote: > > What about the freeswitch.log file when fs is hung? > > > On Jun 28, 2017, at 09:42, Rick Jarvis wrote: > > I’ve tried deleting the db files but no difference unfortunately. Starting > manually or even just restarting the service works, so I believe the XML is > all as it should be. This is my output from systemctl when it’s in this > ‘hung' state: > > ● freeswitch.service - freeswitch > Loaded: loaded (/etc/systemd/system/freeswitch.service; enabled) > Active: active (running) since Wed 2017-06-28 14:36:23 BST; 3min 27s > ago > Process: 452 ExecStart=/usr/bin/freeswitch -ncwait -nonat (code=exited, > status=0/SUCCESS) > Main PID: 472 (freeswitch) > CGroup: /system.slice/freeswitch.service > └─472 /usr/bin/freeswitch -ncwait -nonat > > Jun 28 14:36:20 server systemd[1]: Starting freeswitch... > Jun 28 14:36:20 server freeswitch[452]: 472 Backgrounding. > Jun 28 14:36:23 server freeswitch[452]: FreeSWITCH[452] Waiting for > background process pid:472 to be ready..... > Jun 28 14:36:23 server freeswitch[452]: FreeSWITCH[452] System Ready > pid:472 > Jun 28 14:36:23 server systemd[1]: Started freeswitch. > > > On 28 Jun 2017, at 14:18, Mundkowsky, Robert wrote: > > Yes. I think we had a similar problem in ,I think, two situations. > > 1) The file system database can get messed up. > a. Stop FS > b. You can clear then via rm /export/Apps/freeswitch/var/ > lib/freeswitch/db/* > c. Restart FS > 2) If there is an error in the configuration files. > a. Take a look at /export/Apps/freeswitch/var/ > log/freeswitch/freeswitch.xml.fsxml to see if there are any errors in XML. > > You can get more details of start problem via: > journalctl -xn > systemctl status freeswitch.service > > if you do not get enough details, try starting FS directly > and you might get more details via: > > /…/bin/freeswitch > > > Robert Mundkowsky > > *From:* FreeSWITCH-users [mailto:freeswitch-users- > bounces at lists.freeswitch.org > ] *On Behalf Of *Rick > Jarvis > *Sent:* Wednesday, June 28, 2017 7:44 AM > *To:* FreeSWITCH Users Help > *Subject:* [Freeswitch-users] FS service issues (systemd) > > > Got a weird issue where FS isn’t starting up correctly. Looking at the > logs, it appears just to be attempting to restart SIP profiles (and > failing), and that’s about it. There’s no socket for fs_cli, and a simple > ‘systemctl restart’ fixes it. Anyone had something like this? I’ve tried > fixing it with systemd after / requires, but it hasn’t helped. > > Here’s my .service file (not sure if this is the issue, I'd welcome any > comments / better versions?) > > [Unit] > Description=freeswitch > After=syslog.target network-online.target local-fs.target > > [Service] > ; service > Type=forking > PIDFile=/run/freeswitch/freeswitch.pid > PermissionsStartOnly=true > ExecStart=/usr/bin/freeswitch -ncwait -nonat > ExecStop=/usr/bin/fs_cli -x shutdown > TimeoutSec=45s > Restart=always > ; exec > RuntimeDirectory=freeswitch > RuntimeDirectoryMode=0755 > User=freeswitch > Group=freeswitch > LimitCORE=infinity > LimitNOFILE=100000 > LimitNPROC=60000 > ;LimitSTACK=240 > LimitRTPRIO=infinity > LimitRTTIME=7000000 > IOSchedulingClass=realtime > IOSchedulingPriority=2 > CPUSchedulingPolicy=rr > CPUSchedulingPriority=89 > UMask=0007 > > [Install] > > > > > > > > ------------------------------ > > This e-mail and any files transmitted with it may contain privileged or > confidential information. It is solely for use by the individual for whom > it is intended, even if addressed incorrectly. If you received this e-mail > in error, please notify the sender; do not disclose, copy, distribute, or > take any action in reliance on the contents of this information; and delete > it from your system. Any other use of this e-mail is prohibited. > > Thank you for your compliance. > ------------------------------ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: From rmundkowsky at ets.org Wed Jun 28 13:52:19 2017 From: rmundkowsky at ets.org (Mundkowsky, Robert) Date: Wed, 28 Jun 2017 13:52:19 +0000 Subject: [Freeswitch-users] FS service issues (systemd) In-Reply-To: <518D9916-4387-4C50-9932-DE812A12856B@magicmail.mooo.com> References: <6B172391-E75C-4CA2-A2DE-2C0A652C19EC@magicmail.mooo.com> <518D9916-4387-4C50-9932-DE812A12856B@magicmail.mooo.com> Message-ID: Check the XML anyways, there are some free online validators. Might see if there is any info on these pages that might help: # See https://freeswitch.org/confluence/display/FREESWITCH/Troubleshooting+Debugging # See https://wiki.freeswitch.org/wiki/Troubleshooting_Freeswitch Robert From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rick Jarvis Sent: Wednesday, June 28, 2017 9:43 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] FS service issues (systemd) I’ve tried deleting the db files but no difference unfortunately. Starting manually or even just restarting the service works, so I believe the XML is all as it should be. This is my output from systemctl when it’s in this ‘hung' state: ● freeswitch.service - freeswitch Loaded: loaded (/etc/systemd/system/freeswitch.service; enabled) Active: active (running) since Wed 2017-06-28 14:36:23 BST; 3min 27s ago Process: 452 ExecStart=/usr/bin/freeswitch -ncwait -nonat (code=exited, status=0/SUCCESS) Main PID: 472 (freeswitch) CGroup: /system.slice/freeswitch.service └─472 /usr/bin/freeswitch -ncwait -nonat Jun 28 14:36:20 server systemd[1]: Starting freeswitch... Jun 28 14:36:20 server freeswitch[452]: 472 Backgrounding. Jun 28 14:36:23 server freeswitch[452]: FreeSWITCH[452] Waiting for background process pid:472 to be ready..... Jun 28 14:36:23 server freeswitch[452]: FreeSWITCH[452] System Ready pid:472 Jun 28 14:36:23 server systemd[1]: Started freeswitch. On 28 Jun 2017, at 14:18, Mundkowsky, Robert > wrote: Yes. I think we had a similar problem in ,I think, two situations. 1) The file system database can get messed up. a. Stop FS b. You can clear then via rm /export/Apps/freeswitch/var/lib/freeswitch/db/* c. Restart FS 2) If there is an error in the configuration files. a. Take a look at /export/Apps/freeswitch/var/log/freeswitch/freeswitch.xml.fsxml to see if there are any errors in XML. You can get more details of start problem via: journalctl -xn systemctl status freeswitch.service if you do not get enough details, try starting FS directly and you might get more details via: /…/bin/freeswitch Robert Mundkowsky From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rick Jarvis Sent: Wednesday, June 28, 2017 7:44 AM To: FreeSWITCH Users Help > Subject: [Freeswitch-users] FS service issues (systemd) Got a weird issue where FS isn’t starting up correctly. Looking at the logs, it appears just to be attempting to restart SIP profiles (and failing), and that’s about it. There’s no socket for fs_cli, and a simple ‘systemctl restart’ fixes it. Anyone had something like this? I’ve tried fixing it with systemd after / requires, but it hasn’t helped. Here’s my .service file (not sure if this is the issue, I'd welcome any comments / better versions?) [Unit] Description=freeswitch After=syslog.target network-online.target local-fs.target [Service] ; service Type=forking PIDFile=/run/freeswitch/freeswitch.pid PermissionsStartOnly=true ExecStart=/usr/bin/freeswitch -ncwait -nonat ExecStop=/usr/bin/fs_cli -x shutdown TimeoutSec=45s Restart=always ; exec RuntimeDirectory=freeswitch RuntimeDirectoryMode=0755 User=freeswitch Group=freeswitch LimitCORE=infinity LimitNOFILE=100000 LimitNPROC=60000 ;LimitSTACK=240 LimitRTPRIO=infinity LimitRTTIME=7000000 IOSchedulingClass=realtime IOSchedulingPriority=2 CPUSchedulingPolicy=rr CPUSchedulingPriority=89 UMask=0007 [Install] ________________________________ This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited. Thank you for your compliance. ________________________________ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ________________________________ This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited. Thank you for your compliance. ________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL: From joshua at freeswitch.org Wed Jun 28 14:40:55 2017 From: joshua at freeswitch.org (Joshua Young) Date: Wed, 28 Jun 2017 10:40:55 -0400 Subject: [Freeswitch-users] FS service issues (systemd) In-Reply-To: <6B172391-E75C-4CA2-A2DE-2C0A652C19EC@magicmail.mooo.com> References: <6B172391-E75C-4CA2-A2DE-2C0A652C19EC@magicmail.mooo.com> Message-ID: If you compiled FreeSWITCH, you might want to edit these two lines accordingly. However, if this is a packaged version of FreeSWITCH, this may not be the answer. PIDFile=/run/freeswitch/freeswitch.pid PIDFile=/usr/local/freeswitch/run/freeswitch.pid ExecStart=/usr/bin/freeswitch ExecStart=/usr/local/freeswitch/bin/freeswitch the do the systemctl stuff systemctl daemon-reload systemctl enable freeswitch systemctl start freeswitch On Wed, Jun 28, 2017 at 7:43 AM, Rick Jarvis wrote: > > Got a weird issue where FS isn’t starting up correctly. Looking at the > logs, it appears just to be attempting to restart SIP profiles (and > failing), and that’s about it. There’s no socket for fs_cli, and a simple > ‘systemctl restart’ fixes it. Anyone had something like this? I’ve tried > fixing it with systemd after / requires, but it hasn’t helped. > > Here’s my .service file (not sure if this is the issue, I'd welcome any > comments / better versions?) > > [Unit] > Description=freeswitch > After=syslog.target network-online.target local-fs.target > > [Service] > ; service > Type=forking > PIDFile=/run/freeswitch/freeswitch.pid > PermissionsStartOnly=true > ExecStart=/usr/bin/freeswitch -ncwait -nonat > ExecStop=/usr/bin/fs_cli -x shutdown > TimeoutSec=45s > Restart=always > ; exec > RuntimeDirectory=freeswitch > RuntimeDirectoryMode=0755 > User=freeswitch > Group=freeswitch > LimitCORE=infinity > LimitNOFILE=100000 > LimitNPROC=60000 > ;LimitSTACK=240 > LimitRTPRIO=infinity > LimitRTTIME=7000000 > IOSchedulingClass=realtime > IOSchedulingPriority=2 > CPUSchedulingPolicy=rr > CPUSchedulingPriority=89 > UMask=0007 > > [Install] > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From rick at magicmail.mooo.com Wed Jun 28 14:54:45 2017 From: rick at magicmail.mooo.com (Rick Jarvis) Date: Wed, 28 Jun 2017 15:54:45 +0100 Subject: [Freeswitch-users] FS service issues (systemd) In-Reply-To: References: <6B172391-E75C-4CA2-A2DE-2C0A652C19EC@magicmail.mooo.com> <518D9916-4387-4C50-9932-DE812A12856B@magicmail.mooo.com> <4773C447-BBDD-46BC-8584-4A776A726FF5@gmail.com> <828993D4-5964-4EE7-87B6-D15830E045D3@magicmail.mooo.com> Message-ID: Debian Jessie, FreeSWITCH Version 1.4.26-37~64bit (-37 64bit) It starts fine manually, and works if I restart it with systemctl restart, which is why I’m confused, but leaning towards it being a dependency not having started in time when it boots. I can probably fix it by reinstalling and/or upgrading, I’d just like to know why it’s failing ideally... > On 28 Jun 2017, at 15:44, Giovanni Maruzzelli wrote: > > What exact platform are you using? Os/distro? FS version? Any virtualization? > > Has ever started correctly? > > Have you started it from command line (maybe as root user) and then the problems started to appear? > > Have you tried to reinstall from scratch following strictly the confluence instructions? > > Maybe file/directories permission problem? > > -giovanni > > On 28 June 2017 at 16:32, Rick Jarvis > wrote: > Just looping with failed registrations: > > 2017-06-28 15:29:42.738155 [NOTICE] sofia_reg.c:448 Registering connection1 > 2017-06-28 15:29:42.738155 [NOTICE] sofia_reg.c:448 Registering connection2 > 2017-06-28 15:29:42.738155 [NOTICE] sofia_reg.c:448 Registering xprov > 2017-06-28 15:29:42.738155 [ERR] sofia_reg.c:2392 connection2 Failed Registration with status Service Unavailable [503]. failure #101 > 2017-06-28 15:29:42.738155 [ERR] sofia_reg.c:2392 xprov Failed Registration with status Service Unavailable [503]. failure #101 > 2017-06-28 15:29:42.838117 [ERR] sofia_reg.c:2392 connection1 Failed Registration with status Service Unavailable [503]. failure #101 > 2017-06-28 15:29:43.738108 [WARNING] sofia_reg.c:505 connection1 Failed Registration [503], setting retry to 30 seconds. > 2017-06-28 15:29:43.738108 [WARNING] sofia_reg.c:505 connection2 Failed Registration [503], setting retry to 30 seconds. > 2017-06-28 15:29:43.738108 [WARNING] sofia_reg.c:505 xprov Failed Registration [503], setting retry to 30 seconds. > >> On 28 Jun 2017, at 14:52, Luis Jimenez > wrote: >> >> What about the freeswitch.log file when fs is hung? >> >> >> On Jun 28, 2017, at 09:42, Rick Jarvis > wrote: >> >>> I’ve tried deleting the db files but no difference unfortunately. Starting manually or even just restarting the service works, so I believe the XML is all as it should be. This is my output from systemctl when it’s in this ‘hung' state: >>> >>> ● freeswitch.service - freeswitch >>> Loaded: loaded (/etc/systemd/system/freeswitch.service; enabled) >>> Active: active (running) since Wed 2017-06-28 14:36:23 BST; 3min 27s ago >>> Process: 452 ExecStart=/usr/bin/freeswitch -ncwait -nonat (code=exited, status=0/SUCCESS) >>> Main PID: 472 (freeswitch) >>> CGroup: /system.slice/freeswitch.service >>> └─472 /usr/bin/freeswitch -ncwait -nonat >>> >>> Jun 28 14:36:20 server systemd[1]: Starting freeswitch... >>> Jun 28 14:36:20 server freeswitch[452]: 472 Backgrounding. >>> Jun 28 14:36:23 server freeswitch[452]: FreeSWITCH[452] Waiting for background process pid:472 to be ready..... >>> Jun 28 14:36:23 server freeswitch[452]: FreeSWITCH[452] System Ready pid:472 >>> Jun 28 14:36:23 server systemd[1]: Started freeswitch. >>> >>> >>>> On 28 Jun 2017, at 14:18, Mundkowsky, Robert > wrote: >>>> >>>> Yes. I think we had a similar problem in ,I think, two situations. <> >>>> >>>> 1) The file system database can get messed up. >>>> a. Stop FS >>>> b. You can clear then via rm /export/Apps/freeswitch/var/lib/freeswitch/db/* >>>> c. Restart FS >>>> 2) If there is an error in the configuration files. >>>> a. Take a look at /export/Apps/freeswitch/var/log/freeswitch/freeswitch.xml.fsxml to see if there are any errors in XML. >>>> >>>> You can get more details of start problem via: >>>> journalctl -xn >>>> systemctl status freeswitch.service >>>> >>>> if you do not get enough details, try starting FS directly and you might get more details via: >>>> >>>> /…/bin/freeswitch >>>> >>>> >>>> Robert Mundkowsky >>>> >>>> From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org ] On Behalf Of Rick Jarvis >>>> Sent: Wednesday, June 28, 2017 7:44 AM >>>> To: FreeSWITCH Users Help > >>>> Subject: [Freeswitch-users] FS service issues (systemd) >>>> >>>> >>>> Got a weird issue where FS isn’t starting up correctly. Looking at the logs, it appears just to be attempting to restart SIP profiles (and failing), and that’s about it. There’s no socket for fs_cli, and a simple ‘systemctl restart’ fixes it. Anyone had something like this? I’ve tried fixing it with systemd after / requires, but it hasn’t helped. >>>> >>>> Here’s my .service file (not sure if this is the issue, I'd welcome any comments / better versions?) >>>> >>>> [Unit] >>>> Description=freeswitch >>>> After=syslog.target network-online.target local-fs.target >>>> >>>> [Service] >>>> ; service >>>> Type=forking >>>> PIDFile=/run/freeswitch/freeswitch.pid >>>> PermissionsStartOnly=true >>>> ExecStart=/usr/bin/freeswitch -ncwait -nonat >>>> ExecStop=/usr/bin/fs_cli -x shutdown >>>> TimeoutSec=45s >>>> Restart=always >>>> ; exec >>>> RuntimeDirectory=freeswitch >>>> RuntimeDirectoryMode=0755 >>>> User=freeswitch >>>> Group=freeswitch >>>> LimitCORE=infinity >>>> LimitNOFILE=100000 >>>> LimitNPROC=60000 >>>> ;LimitSTACK=240 >>>> LimitRTPRIO=infinity >>>> LimitRTTIME=7000000 >>>> IOSchedulingClass=realtime >>>> IOSchedulingPriority=2 >>>> CPUSchedulingPolicy=rr >>>> CPUSchedulingPriority=89 >>>> UMask=0007 >>>> >>>> [Install] >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited. >>>> >>>> >>>> Thank you for your compliance. >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 > OpenTelecom.IT > cell: +39 347 266 56 18 > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From gmaruzz at gmail.com Wed Jun 28 14:57:37 2017 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Wed, 28 Jun 2017 16:57:37 +0200 Subject: [Freeswitch-users] FS service issues (systemd) In-Reply-To: References: <6B172391-E75C-4CA2-A2DE-2C0A652C19EC@magicmail.mooo.com> <518D9916-4387-4C50-9932-DE812A12856B@magicmail.mooo.com> <4773C447-BBDD-46BC-8584-4A776A726FF5@gmail.com> <828993D4-5964-4EE7-87B6-D15830E045D3@magicmail.mooo.com> Message-ID: On 28 June 2017 at 16:54, Rick Jarvis wrote: > Debian Jessie, FreeSWITCH Version 1.4.26-37~64bit (-37 64bit) > > It starts fine manually, and works if I restart it with systemctl restart, > which is why I’m confused, but leaning towards it being a dependency not > having started in time when it boots. I can probably fix it by reinstalling > and/or upgrading, I’d just like to know why it’s failing ideally... > Use latest stable: 1.6.18, at this moment, and reinstall fron scratch, on a fresh new Debian Jessie install -giovanni > > > On 28 Jun 2017, at 15:44, Giovanni Maruzzelli wrote: > > What exact platform are you using? Os/distro? FS version? Any > virtualization? > > Has ever started correctly? > > Have you started it from command line (maybe as root user) and then the > problems started to appear? > > Have you tried to reinstall from scratch following strictly the confluence > instructions? > > Maybe file/directories permission problem? > > -giovanni > > On 28 June 2017 at 16:32, Rick Jarvis wrote: > >> Just looping with failed registrations: >> >> 2017-06-28 15:29:42.738155 [NOTICE] sofia_reg.c:448 Registering >> connection1 >> 2017-06-28 15:29:42.738155 [NOTICE] sofia_reg.c:448 Registering >> connection2 >> 2017-06-28 15:29:42.738155 [NOTICE] sofia_reg.c:448 Registering xprov >> 2017-06-28 15:29:42.738155 [ERR] sofia_reg.c:2392 connection2 Failed >> Registration with status Service Unavailable [503]. failure #101 >> 2017-06-28 15:29:42.738155 [ERR] sofia_reg.c:2392 xprov Failed >> Registration with status Service Unavailable [503]. failure #101 >> 2017-06-28 15:29:42.838117 [ERR] sofia_reg.c:2392 connection1 Failed >> Registration with status Service Unavailable [503]. failure #101 >> 2017-06-28 15:29:43.738108 [WARNING] sofia_reg.c:505 connection1 Failed >> Registration [503], setting retry to 30 seconds. >> 2017-06-28 15:29:43.738108 [WARNING] sofia_reg.c:505 connection2 Failed >> Registration [503], setting retry to 30 seconds. >> 2017-06-28 15:29:43.738108 [WARNING] sofia_reg.c:505 xprov Failed >> Registration [503], setting retry to 30 seconds. >> >> On 28 Jun 2017, at 14:52, Luis Jimenez wrote: >> >> What about the freeswitch.log file when fs is hung? >> >> >> On Jun 28, 2017, at 09:42, Rick Jarvis wrote: >> >> I’ve tried deleting the db files but no difference unfortunately. >> Starting manually or even just restarting the service works, so I believe >> the XML is all as it should be. This is my output from systemctl when it’s >> in this ‘hung' state: >> >> ● freeswitch.service - freeswitch >> Loaded: loaded (/etc/systemd/system/freeswitch.service; enabled) >> Active: active (running) since Wed 2017-06-28 14:36:23 BST; 3min 27s >> ago >> Process: 452 ExecStart=/usr/bin/freeswitch -ncwait -nonat (code=exited, >> status=0/SUCCESS) >> Main PID: 472 (freeswitch) >> CGroup: /system.slice/freeswitch.service >> └─472 /usr/bin/freeswitch -ncwait -nonat >> >> Jun 28 14:36:20 server systemd[1]: Starting freeswitch... >> Jun 28 14:36:20 server freeswitch[452]: 472 Backgrounding. >> Jun 28 14:36:23 server freeswitch[452]: FreeSWITCH[452] Waiting for >> background process pid:472 to be ready..... >> Jun 28 14:36:23 server freeswitch[452]: FreeSWITCH[452] System Ready >> pid:472 >> Jun 28 14:36:23 server systemd[1]: Started freeswitch. >> >> >> On 28 Jun 2017, at 14:18, Mundkowsky, Robert wrote: >> >> Yes. I think we had a similar problem in ,I think, two situations. >> >> 1) The file system database can get messed up. >> a. Stop FS >> b. You can clear then via rm /export/Apps/freeswitch/var/li >> b/freeswitch/db/* >> c. Restart FS >> 2) If there is an error in the configuration files. >> a. Take a look at /export/Apps/freeswitch/var/lo >> g/freeswitch/freeswitch.xml.fsxml to see if there are any errors in XML. >> >> You can get more details of start problem via: >> journalctl -xn >> systemctl status freeswitch.service >> >> if you do not get enough details, try starting FS >> directly and you might get more details via: >> >> /…/bin/freeswitch >> >> >> Robert Mundkowsky >> >> *From:* FreeSWITCH-users [mailto:freeswitch-users-bounc >> es at lists.freeswitch.org ] *On >> Behalf Of *Rick Jarvis >> *Sent:* Wednesday, June 28, 2017 7:44 AM >> *To:* FreeSWITCH Users Help >> *Subject:* [Freeswitch-users] FS service issues (systemd) >> >> >> Got a weird issue where FS isn’t starting up correctly. Looking at the >> logs, it appears just to be attempting to restart SIP profiles (and >> failing), and that’s about it. There’s no socket for fs_cli, and a simple >> ‘systemctl restart’ fixes it. Anyone had something like this? I’ve tried >> fixing it with systemd after / requires, but it hasn’t helped. >> >> Here’s my .service file (not sure if this is the issue, I'd welcome any >> comments / better versions?) >> >> [Unit] >> Description=freeswitch >> After=syslog.target network-online.target local-fs.target >> >> [Service] >> ; service >> Type=forking >> PIDFile=/run/freeswitch/freeswitch.pid >> PermissionsStartOnly=true >> ExecStart=/usr/bin/freeswitch -ncwait -nonat >> ExecStop=/usr/bin/fs_cli -x shutdown >> TimeoutSec=45s >> Restart=always >> ; exec >> RuntimeDirectory=freeswitch >> RuntimeDirectoryMode=0755 >> User=freeswitch >> Group=freeswitch >> LimitCORE=infinity >> LimitNOFILE=100000 >> LimitNPROC=60000 >> ;LimitSTACK=240 >> LimitRTPRIO=infinity >> LimitRTTIME=7000000 >> IOSchedulingClass=realtime >> IOSchedulingPriority=2 >> CPUSchedulingPolicy=rr >> CPUSchedulingPriority=89 >> UMask=0007 >> >> [Install] >> >> >> >> >> >> >> >> ------------------------------ >> >> This e-mail and any files transmitted with it may contain privileged or >> confidential information. It is solely for use by the individual for whom >> it is intended, even if addressed incorrectly. If you received this e-mail >> in error, please notify the sender; do not disclose, copy, distribute, or >> take any action in reliance on the contents of this information; and delete >> it from your system. Any other use of this e-mail is prohibited. >> >> Thank you for your compliance. >> ------------------------------ >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 > OpenTelecom.IT > cell: +39 347 266 56 18 > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at 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 at 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 at 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 OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: From rmundkowsky at ets.org Wed Jun 28 14:58:30 2017 From: rmundkowsky at ets.org (Mundkowsky, Robert) Date: Wed, 28 Jun 2017 14:58:30 +0000 Subject: [Freeswitch-users] FS service issues (systemd) In-Reply-To: References: <6B172391-E75C-4CA2-A2DE-2C0A652C19EC@magicmail.mooo.com> Message-ID: Might check that PID file is deleted before you start FS and that there are no FS trying to start already. Robert Mundkowsky From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Joshua Young Sent: Wednesday, June 28, 2017 10:41 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] FS service issues (systemd) If you compiled FreeSWITCH, you might want to edit these two lines accordingly. However, if this is a packaged version of FreeSWITCH, this may not be the answer. PIDFile=/run/freeswitch/freeswitch.pid PIDFile=/usr/local/freeswitch/run/freeswitch.pid ExecStart=/usr/bin/freeswitch ExecStart=/usr/local/freeswitch/bin/freeswitch the do the systemctl stuff systemctl daemon-reload systemctl enable freeswitch systemctl start freeswitch On Wed, Jun 28, 2017 at 7:43 AM, Rick Jarvis > wrote: Got a weird issue where FS isn’t starting up correctly. Looking at the logs, it appears just to be attempting to restart SIP profiles (and failing), and that’s about it. There’s no socket for fs_cli, and a simple ‘systemctl restart’ fixes it. Anyone had something like this? I’ve tried fixing it with systemd after / requires, but it hasn’t helped. Here’s my .service file (not sure if this is the issue, I'd welcome any comments / better versions?) [Unit] Description=freeswitch After=syslog.target network-online.target local-fs.target [Service] ; service Type=forking PIDFile=/run/freeswitch/freeswitch.pid PermissionsStartOnly=true ExecStart=/usr/bin/freeswitch -ncwait -nonat ExecStop=/usr/bin/fs_cli -x shutdown TimeoutSec=45s Restart=always ; exec RuntimeDirectory=freeswitch RuntimeDirectoryMode=0755 User=freeswitch Group=freeswitch LimitCORE=infinity LimitNOFILE=100000 LimitNPROC=60000 ;LimitSTACK=240 LimitRTPRIO=infinity LimitRTTIME=7000000 IOSchedulingClass=realtime IOSchedulingPriority=2 CPUSchedulingPolicy=rr CPUSchedulingPriority=89 UMask=0007 [Install] _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ________________________________ This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited. Thank you for your compliance. ________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL: From rmundkowsky at ets.org Wed Jun 28 15:11:07 2017 From: rmundkowsky at ets.org (Mundkowsky, Robert) Date: Wed, 28 Jun 2017 15:11:07 +0000 Subject: [Freeswitch-users] FS service issues (systemd) In-Reply-To: References: <6B172391-E75C-4CA2-A2DE-2C0A652C19EC@magicmail.mooo.com> <518D9916-4387-4C50-9932-DE812A12856B@magicmail.mooo.com> <4773C447-BBDD-46BC-8584-4A776A726FF5@gmail.com> <828993D4-5964-4EE7-87B6-D15830E045D3@magicmail.mooo.com> Message-ID: Few more suggestions: 1 Make sure run and log directories are owned by same user that you run it as. 2 The newer version of FS are more stable for us. Robert Mundkowsky From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Giovanni Maruzzelli Sent: Wednesday, June 28, 2017 10:58 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] FS service issues (systemd) On 28 June 2017 at 16:54, Rick Jarvis > wrote: Debian Jessie, FreeSWITCH Version 1.4.26-37~64bit (-37 64bit) It starts fine manually, and works if I restart it with systemctl restart, which is why I’m confused, but leaning towards it being a dependency not having started in time when it boots. I can probably fix it by reinstalling and/or upgrading, I’d just like to know why it’s failing ideally... Use latest stable: 1.6.18, at this moment, and reinstall fron scratch, on a fresh new Debian Jessie install -giovanni On 28 Jun 2017, at 15:44, Giovanni Maruzzelli > wrote: What exact platform are you using? Os/distro? FS version? Any virtualization? Has ever started correctly? Have you started it from command line (maybe as root user) and then the problems started to appear? Have you tried to reinstall from scratch following strictly the confluence instructions? Maybe file/directories permission problem? -giovanni On 28 June 2017 at 16:32, Rick Jarvis > wrote: Just looping with failed registrations: 2017-06-28 15:29:42.738155 [NOTICE] sofia_reg.c:448 Registering connection1 2017-06-28 15:29:42.738155 [NOTICE] sofia_reg.c:448 Registering connection2 2017-06-28 15:29:42.738155 [NOTICE] sofia_reg.c:448 Registering xprov 2017-06-28 15:29:42.738155 [ERR] sofia_reg.c:2392 connection2 Failed Registration with status Service Unavailable [503]. failure #101 2017-06-28 15:29:42.738155 [ERR] sofia_reg.c:2392 xprov Failed Registration with status Service Unavailable [503]. failure #101 2017-06-28 15:29:42.838117 [ERR] sofia_reg.c:2392 connection1 Failed Registration with status Service Unavailable [503]. failure #101 2017-06-28 15:29:43.738108 [WARNING] sofia_reg.c:505 connection1 Failed Registration [503], setting retry to 30 seconds. 2017-06-28 15:29:43.738108 [WARNING] sofia_reg.c:505 connection2 Failed Registration [503], setting retry to 30 seconds. 2017-06-28 15:29:43.738108 [WARNING] sofia_reg.c:505 xprov Failed Registration [503], setting retry to 30 seconds. On 28 Jun 2017, at 14:52, Luis Jimenez > wrote: What about the freeswitch.log file when fs is hung? On Jun 28, 2017, at 09:42, Rick Jarvis > wrote: I’ve tried deleting the db files but no difference unfortunately. Starting manually or even just restarting the service works, so I believe the XML is all as it should be. This is my output from systemctl when it’s in this ‘hung' state: ● freeswitch.service - freeswitch Loaded: loaded (/etc/systemd/system/freeswitch.service; enabled) Active: active (running) since Wed 2017-06-28 14:36:23 BST; 3min 27s ago Process: 452 ExecStart=/usr/bin/freeswitch -ncwait -nonat (code=exited, status=0/SUCCESS) Main PID: 472 (freeswitch) CGroup: /system.slice/freeswitch.service └─472 /usr/bin/freeswitch -ncwait -nonat Jun 28 14:36:20 server systemd[1]: Starting freeswitch... Jun 28 14:36:20 server freeswitch[452]: 472 Backgrounding. Jun 28 14:36:23 server freeswitch[452]: FreeSWITCH[452] Waiting for background process pid:472 to be ready..... Jun 28 14:36:23 server freeswitch[452]: FreeSWITCH[452] System Ready pid:472 Jun 28 14:36:23 server systemd[1]: Started freeswitch. On 28 Jun 2017, at 14:18, Mundkowsky, Robert > wrote: Yes. I think we had a similar problem in ,I think, two situations. 1) The file system database can get messed up. a. Stop FS b. You can clear then via rm /export/Apps/freeswitch/var/lib/freeswitch/db/* c. Restart FS 2) If there is an error in the configuration files. a. Take a look at /export/Apps/freeswitch/var/log/freeswitch/freeswitch.xml.fsxml to see if there are any errors in XML. You can get more details of start problem via: journalctl -xn systemctl status freeswitch.service if you do not get enough details, try starting FS directly and you might get more details via: /…/bin/freeswitch Robert Mundkowsky From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rick Jarvis Sent: Wednesday, June 28, 2017 7:44 AM To: FreeSWITCH Users Help > Subject: [Freeswitch-users] FS service issues (systemd) Got a weird issue where FS isn’t starting up correctly. Looking at the logs, it appears just to be attempting to restart SIP profiles (and failing), and that’s about it. There’s no socket for fs_cli, and a simple ‘systemctl restart’ fixes it. Anyone had something like this? I’ve tried fixing it with systemd after / requires, but it hasn’t helped. Here’s my .service file (not sure if this is the issue, I'd welcome any comments / better versions?) [Unit] Description=freeswitch After=syslog.target network-online.target local-fs.target [Service] ; service Type=forking PIDFile=/run/freeswitch/freeswitch.pid PermissionsStartOnly=true ExecStart=/usr/bin/freeswitch -ncwait -nonat ExecStop=/usr/bin/fs_cli -x shutdown TimeoutSec=45s Restart=always ; exec RuntimeDirectory=freeswitch RuntimeDirectoryMode=0755 User=freeswitch Group=freeswitch LimitCORE=infinity LimitNOFILE=100000 LimitNPROC=60000 ;LimitSTACK=240 LimitRTPRIO=infinity LimitRTTIME=7000000 IOSchedulingClass=realtime IOSchedulingPriority=2 CPUSchedulingPolicy=rr CPUSchedulingPriority=89 UMask=0007 [Install] ________________________________ This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited. Thank you for your compliance. ________________________________ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 OpenTelecom.IT cell: +39 347 266 56 18 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at 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 at 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 at 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 at 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 OpenTelecom.IT cell: +39 347 266 56 18 ________________________________ This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited. Thank you for your compliance. ________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL: From nbhatti at gmail.com Wed Jun 28 16:48:10 2017 From: nbhatti at gmail.com (Muhammad Naseer Bhatti) Date: Wed, 28 Jun 2017 12:48:10 -0400 Subject: [Freeswitch-users] How to delay setInputCallback() while playing audio? In-Reply-To: References: Message-ID: Or something like sched_api_function() which calls the function to detect events? -- Sent with Airmail From: Muhammad Naseer Bhatti Reply: Muhammad Naseer Bhatti Date: June 28, 2017 at 12:18:40 AM To: FreeSWITCH Users Help , Brian West Subject: Re: [Freeswitch-users] How to delay setInputCallback() while playing audio? playbackMessage() function is actually doing session:streamFile(audio_file). Input callback is functioning fine, but I want to delay detection a few seconds somehow. On the other hand, is there a way to stop a running session:streamFile() ? -- Sent with Airmail From: Brian West Reply: FreeSWITCH Users Help Date: June 28, 2017 at 12:15:20 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to delay setInputCallback() while playing audio? Show us how you're doing the playback, If you're doing session:execute('playback', 'blah.wav'); The input callback won't function you have to use session:streamFile instead. /b On Tue, Jun 27, 2017 at 1:10 PM, Muhammad Naseer Bhatti wrote: > I am running an IVR which collects DTMF input while the file is being > played, written in Lua. I set InputCallback and then listen for DTMF > events. If event found, take actions etc. > > session:setInputCallback("eventHandler") > playbackMessage (playback_delay) > > Facing a strange issues with Verizon where they send a quick RTP Event > write after starting the audio, which triggers the DTMF event and bad > things happen in my script. This only happens within first 1 to 2 seconds > of RTP. How can I have my InputCallback listen for events only after a few > seconds while still playing the audio? or do I have to switch to > playAndGetDigits() ? > > > -- > > Sent with Airmail > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Book a phone call (CST) Allison prompts for FreeSWITCH: *https://www.gofundme.com/allison-prompts-for-freeswitch* Got Bugs? Report them here ! | Reddit: /r/freeswitch *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *Skype:*briankwest _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From brian at freeswitch.org Wed Jun 28 17:11:59 2017 From: brian at freeswitch.org (Brian West) Date: Wed, 28 Jun 2017 12:11:59 -0500 Subject: [Freeswitch-users] How to delay setInputCallback() while playing audio? In-Reply-To: References: Message-ID: use streamFile silence_stream://2000 On Wed, Jun 28, 2017 at 11:48 AM, Muhammad Naseer Bhatti wrote: > > Or something like sched_api_function() which calls the function to detect > events? > > -- > > Sent with Airmail > > From: Muhammad Naseer Bhatti > Reply: Muhammad Naseer Bhatti > Date: June 28, 2017 at 12:18:40 AM > To: FreeSWITCH Users Help > , Brian West > > > Subject: Re: [Freeswitch-users] How to delay setInputCallback() while > playing audio? > > > playbackMessage() function is actually doing session:streamFile(audio_file). > Input callback is functioning fine, but I want to delay detection a few > seconds somehow. > > On the other hand, is there a way to stop a running session:streamFile() ? > > -- > > Sent with Airmail > > From: Brian West > Reply: FreeSWITCH Users Help > > Date: June 28, 2017 at 12:15:20 AM > To: FreeSWITCH Users Help > > Subject: Re: [Freeswitch-users] How to delay setInputCallback() while > playing audio? > > Show us how you're doing the playback, If you're doing > session:execute('playback', 'blah.wav'); The input callback won't function > you have to use session:streamFile instead. > > /b > > > On Tue, Jun 27, 2017 at 1:10 PM, Muhammad Naseer Bhatti > wrote: > >> I am running an IVR which collects DTMF input while the file is being >> played, written in Lua. I set InputCallback and then listen for DTMF >> events. If event found, take actions etc. >> >> session:setInputCallback("eventHandler") >> playbackMessage (playback_delay) >> >> Facing a strange issues with Verizon where they send a quick RTP Event >> write after starting the audio, which triggers the DTMF event and bad >> things happen in my script. This only happens within first 1 to 2 seconds >> of RTP. How can I have my InputCallback listen for events only after a few >> seconds while still playing the audio? or do I have to switch to >> playAndGetDigits() ? >> >> >> -- >> >> Sent with Airmail >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > > *Brian West* > brian at freeswitch.org > > *Twitter: @FreeSWITCH , @briankwest* > > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > Book a phone call (CST) > > Allison prompts for FreeSWITCH: > > *https://www.gofundme.com/allison-prompts-for-freeswitch* > > > Got Bugs? Report them here ! | Reddit: > /r/freeswitch > > *T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002 <(918)%20420-9002> > | *M:*+1918424WEST (9378) > *Skype:*briankwest > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Book a phone call (CST) Allison prompts for FreeSWITCH: *https://www.gofundme.com/allison-prompts-for-freeswitch* Got Bugs? Report them here ! | Reddit: /r/freeswitch *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: From wgaitner at gmail.com Wed Jun 28 16:48:57 2017 From: wgaitner at gmail.com (Vladimir Gaitner) Date: Wed, 28 Jun 2017 19:48:57 +0300 Subject: [Freeswitch-users] SKYPE: mod_skypopen END-OF-LIFE Message-ID: Hello Geowanni, Do you have any planns to rewrite the module? -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Wed Jun 28 17:16:04 2017 From: mike at jerris.com (Michael Jerris) Date: Wed, 28 Jun 2017 13:16:04 -0400 Subject: [Freeswitch-users] SKYPE: mod_skypopen END-OF-LIFE In-Reply-To: References: Message-ID: Currently Skype provides no API to make rewriting possible. If one is released, we would look at it at that time. Mike > On Jun 28, 2017, at 12:48 PM, Vladimir Gaitner wrote: > > Hello Geowanni, > Do you have any planns to rewrite the module? > From vbvbrj at gmail.com Wed Jun 28 17:17:16 2017 From: vbvbrj at gmail.com (Mimiko) Date: Wed, 28 Jun 2017 20:17:16 +0300 Subject: [Freeswitch-users] tier change event Message-ID: Hello. mod_callcenter generate agent's state or status change events. Is there events generated when a tier is added, changed, removed? From nbhatti at gmail.com Wed Jun 28 18:02:26 2017 From: nbhatti at gmail.com (Muhammad Naseer Bhatti) Date: Wed, 28 Jun 2017 14:02:26 -0400 Subject: [Freeswitch-users] How to delay setInputCallback() while playing audio? In-Reply-To: References: Message-ID: I guess this would only introduce a silence of 2 seconds. I am trying to play the file but delay DTMF detection for a few seconds. So my file is being played but I want to ignore a few seconds of DTMF inputs for callback function. Can’t think of a better way to do that. -- Sent with Airmail From: Brian West Reply: Brian West Date: June 28, 2017 at 10:12:00 PM To: Muhammad Naseer Bhatti Cc: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to delay setInputCallback() while playing audio? use streamFile silence_stream://2000 On Wed, Jun 28, 2017 at 11:48 AM, Muhammad Naseer Bhatti wrote: > > Or something like sched_api_function() which calls the function to detect > events? > > -- > > Sent with Airmail > > From: Muhammad Naseer Bhatti > Reply: Muhammad Naseer Bhatti > Date: June 28, 2017 at 12:18:40 AM > To: FreeSWITCH Users Help > , Brian West > > > Subject: Re: [Freeswitch-users] How to delay setInputCallback() while > playing audio? > > > playbackMessage() function is actually doing session:streamFile(audio_file). > Input callback is functioning fine, but I want to delay detection a few > seconds somehow. > > On the other hand, is there a way to stop a running session:streamFile() ? > > -- > > Sent with Airmail > > From: Brian West > Reply: FreeSWITCH Users Help > > Date: June 28, 2017 at 12:15:20 AM > To: FreeSWITCH Users Help > > Subject: Re: [Freeswitch-users] How to delay setInputCallback() while > playing audio? > > Show us how you're doing the playback, If you're doing > session:execute('playback', 'blah.wav'); The input callback won't function > you have to use session:streamFile instead. > > /b > > > On Tue, Jun 27, 2017 at 1:10 PM, Muhammad Naseer Bhatti > wrote: > >> I am running an IVR which collects DTMF input while the file is being >> played, written in Lua. I set InputCallback and then listen for DTMF >> events. If event found, take actions etc. >> >> session:setInputCallback("eventHandler") >> playbackMessage (playback_delay) >> >> Facing a strange issues with Verizon where they send a quick RTP Event >> write after starting the audio, which triggers the DTMF event and bad >> things happen in my script. This only happens within first 1 to 2 seconds >> of RTP. How can I have my InputCallback listen for events only after a few >> seconds while still playing the audio? or do I have to switch to >> playAndGetDigits() ? >> >> >> -- >> >> Sent with Airmail >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > > *Brian West* > brian at freeswitch.org > > *Twitter: @FreeSWITCH , @briankwest* > > http://www.freeswitchbook.com > http://www.freeswitchcookbook.com > > Book a phone call (CST) > > Allison prompts for FreeSWITCH: > > *https://www.gofundme.com/allison-prompts-for-freeswitch* > > > Got Bugs? Report them here ! | Reddit: > /r/freeswitch > > *T:*+19184209001 <(918)%20420-9001> | *F:*+19184209002 <(918)%20420-9002> > | *M:*+1918424WEST (9378) > *Skype:*briankwest > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- *Brian West* brian at freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Book a phone call (CST) Allison prompts for FreeSWITCH: *https://www.gofundme.com/allison-prompts-for-freeswitch* Got Bugs? Report them here ! | Reddit: /r/freeswitch *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *Skype:*briankwest -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Wed Jun 28 18:12:16 2017 From: mike at jerris.com (Michael Jerris) Date: Wed, 28 Jun 2017 14:12:16 -0400 Subject: [Freeswitch-users] How to delay setInputCallback() while playing audio? In-Reply-To: References: Message-ID: <86EB8621-BBF2-49F8-83BC-A139007D9E32@jerris.com> you’d have to handle ignoring for a couple seconds in your callback.. there is no way to delay setting the callback that I can think of. > On Jun 28, 2017, at 2:02 PM, Muhammad Naseer Bhatti wrote: > > > I guess this would only introduce a silence of 2 seconds. I am trying to play the file but delay DTMF detection for a few seconds. So my file is being played but I want to ignore a few seconds of DTMF inputs for callback function. Can’t think of a better way to do that. > > -- > > Sent with Airmail > > From: Brian West > Reply: Brian West > Date: June 28, 2017 at 10:12:00 PM > To: Muhammad Naseer Bhatti > Cc: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] How to delay setInputCallback() while playing audio? > >> use streamFile silence_stream://2000 >> >> >> >> On Wed, Jun 28, 2017 at 11:48 AM, Muhammad Naseer Bhatti > wrote: >> >> Or something like sched_api_function() which calls the function to detect events? >> >> -- >> >> Sent with Airmail >> >> From: Muhammad Naseer Bhatti >> Reply: Muhammad Naseer Bhatti >> Date: June 28, 2017 at 12:18:40 AM >> To: FreeSWITCH Users Help , Brian West >> >> Subject: Re: [Freeswitch-users] How to delay setInputCallback() while playing audio? >> >>> >>> playbackMessage() function is actually doing session:streamFile(audio_file). Input callback is functioning fine, but I want to delay detection a few seconds somehow. >>> >>> On the other hand, is there a way to stop a running session:streamFile() ? >>> >>> -- >>> >>> Sent with Airmail >>> >>> From: Brian West >>> Reply: FreeSWITCH Users Help >>> Date: June 28, 2017 at 12:15:20 AM >>> To: FreeSWITCH Users Help >>> Subject: Re: [Freeswitch-users] How to delay setInputCallback() while playing audio? >>> >>>> Show us how you're doing the playback, If you're doing session:execute('playback', 'blah.wav'); The input callback won't function you have to use session:streamFile instead. >>>> >>>> /b >>>> >>>> >>>> On Tue, Jun 27, 2017 at 1:10 PM, Muhammad Naseer Bhatti > wrote: >>>> I am running an IVR which collects DTMF input while the file is being played, written in Lua. I set InputCallback and then listen for DTMF events. If event found, take actions etc. >>>> >>>> session:setInputCallback("eventHandler") >>>> playbackMessage (playback_delay) >>>> >>>> Facing a strange issues with Verizon where they send a quick RTP Event write after starting the audio, which triggers the DTMF event and bad things happen in my script. This only happens within first 1 to 2 seconds of RTP. How can I have my InputCallback listen for events only after a few seconds while still playing the audio? or do I have to switch to playAndGetDigits() ? >>>> >>>> >>>> -- -------------- next part -------------- An HTML attachment was scrubbed... URL: From rick at magicmail.mooo.com Wed Jun 28 18:52:55 2017 From: rick at magicmail.mooo.com (Rick Jarvis) Date: Wed, 28 Jun 2017 19:52:55 +0100 Subject: [Freeswitch-users] FS service issues (systemd) In-Reply-To: References: <6B172391-E75C-4CA2-A2DE-2C0A652C19EC@magicmail.mooo.com> <518D9916-4387-4C50-9932-DE812A12856B@magicmail.mooo.com> <4773C447-BBDD-46BC-8584-4A776A726FF5@gmail.com> <828993D4-5964-4EE7-87B6-D15830E045D3@magicmail.mooo.com> Message-ID: <1A108FA3-C445-48C4-A3AA-98D4B4A2B275@magicmail.mooo.com> Hmm, wondering why the deb repo is still dishing out 1.4? :( > On 28 Jun 2017, at 16:11, Mundkowsky, Robert wrote: > > Few more suggestions: <> > > 1 Make sure run and log directories are owned by same user that you run it as. > 2 The newer version of FS are more stable for us. > > Robert Mundkowsky > > From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org ] On Behalf Of Giovanni Maruzzelli > Sent: Wednesday, June 28, 2017 10:58 AM > To: FreeSWITCH Users Help > > Subject: Re: [Freeswitch-users] FS service issues (systemd) > > > > On 28 June 2017 at 16:54, Rick Jarvis > wrote: > Debian Jessie, FreeSWITCH Version 1.4.26-37~64bit (-37 64bit) > > It starts fine manually, and works if I restart it with systemctl restart, which is why I’m confused, but leaning towards it being a dependency not having started in time when it boots. I can probably fix it by reinstalling and/or upgrading, I’d just like to know why it’s failing ideally... > > Use latest stable: 1.6.18, at this moment, and reinstall fron scratch, on a fresh new Debian Jessie install > > -giovanni > > > > > On 28 Jun 2017, at 15:44, Giovanni Maruzzelli > wrote: > > What exact platform are you using? Os/distro? FS version? Any virtualization? > > Has ever started correctly? > > Have you started it from command line (maybe as root user) and then the problems started to appear? > > Have you tried to reinstall from scratch following strictly the confluence instructions? > > Maybe file/directories permission problem? > > -giovanni > > On 28 June 2017 at 16:32, Rick Jarvis > wrote: > Just looping with failed registrations: > > 2017-06-28 15:29:42.738155 [NOTICE] sofia_reg.c:448 Registering connection1 > 2017-06-28 15:29:42.738155 [NOTICE] sofia_reg.c:448 Registering connection2 > 2017-06-28 15:29:42.738155 [NOTICE] sofia_reg.c:448 Registering xprov > 2017-06-28 15:29:42.738155 [ERR] sofia_reg.c:2392 connection2 Failed Registration with status Service Unavailable [503]. failure #101 > 2017-06-28 15:29:42.738155 [ERR] sofia_reg.c:2392 xprov Failed Registration with status Service Unavailable [503]. failure #101 > 2017-06-28 15:29:42.838117 [ERR] sofia_reg.c:2392 connection1 Failed Registration with status Service Unavailable [503]. failure #101 > 2017-06-28 15:29:43.738108 [WARNING] sofia_reg.c:505 connection1 Failed Registration [503], setting retry to 30 seconds. > 2017-06-28 15:29:43.738108 [WARNING] sofia_reg.c:505 connection2 Failed Registration [503], setting retry to 30 seconds. > 2017-06-28 15:29:43.738108 [WARNING] sofia_reg.c:505 xprov Failed Registration [503], setting retry to 30 seconds. > > On 28 Jun 2017, at 14:52, Luis Jimenez > wrote: > > What about the freeswitch.log file when fs is hung? > > > On Jun 28, 2017, at 09:42, Rick Jarvis > wrote: > > I’ve tried deleting the db files but no difference unfortunately. Starting manually or even just restarting the service works, so I believe the XML is all as it should be. This is my output from systemctl when it’s in this ‘hung' state: > > ● freeswitch.service - freeswitch > Loaded: loaded (/etc/systemd/system/freeswitch.service; enabled) > Active: active (running) since Wed 2017-06-28 14:36:23 BST; 3min 27s ago > Process: 452 ExecStart=/usr/bin/freeswitch -ncwait -nonat (code=exited, status=0/SUCCESS) > Main PID: 472 (freeswitch) > CGroup: /system.slice/freeswitch.service > └─472 /usr/bin/freeswitch -ncwait -nonat > > Jun 28 14:36:20 server systemd[1]: Starting freeswitch... > Jun 28 14:36:20 server freeswitch[452]: 472 Backgrounding. > Jun 28 14:36:23 server freeswitch[452]: FreeSWITCH[452] Waiting for background process pid:472 to be ready..... > Jun 28 14:36:23 server freeswitch[452]: FreeSWITCH[452] System Ready pid:472 > Jun 28 14:36:23 server systemd[1]: Started freeswitch. > > > On 28 Jun 2017, at 14:18, Mundkowsky, Robert > wrote: > > Yes. I think we had a similar problem in ,I think, two situations. <> > > 1) The file system database can get messed up. > a. Stop FS > b. You can clear then via rm /export/Apps/freeswitch/var/lib/freeswitch/db/* > c. Restart FS > 2) If there is an error in the configuration files. > a. Take a look at /export/Apps/freeswitch/var/log/freeswitch/freeswitch.xml.fsxml to see if there are any errors in XML. > > You can get more details of start problem via: > journalctl -xn > systemctl status freeswitch.service > > if you do not get enough details, try starting FS directly and you might get more details via: > > /…/bin/freeswitch > > > Robert Mundkowsky > > From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org ] On Behalf Of Rick Jarvis > Sent: Wednesday, June 28, 2017 7:44 AM > To: FreeSWITCH Users Help > > Subject: [Freeswitch-users] FS service issues (systemd) > > > Got a weird issue where FS isn’t starting up correctly. Looking at the logs, it appears just to be attempting to restart SIP profiles (and failing), and that’s about it. There’s no socket for fs_cli, and a simple ‘systemctl restart’ fixes it. Anyone had something like this? I’ve tried fixing it with systemd after / requires, but it hasn’t helped. > > Here’s my .service file (not sure if this is the issue, I'd welcome any comments / better versions?) > > [Unit] > Description=freeswitch > After=syslog.target network-online.target local-fs.target > > [Service] > ; service > Type=forking > PIDFile=/run/freeswitch/freeswitch.pid > PermissionsStartOnly=true > ExecStart=/usr/bin/freeswitch -ncwait -nonat > ExecStop=/usr/bin/fs_cli -x shutdown > TimeoutSec=45s > Restart=always > ; exec > RuntimeDirectory=freeswitch > RuntimeDirectoryMode=0755 > User=freeswitch > Group=freeswitch > LimitCORE=infinity > LimitNOFILE=100000 > LimitNPROC=60000 > ;LimitSTACK=240 > LimitRTPRIO=infinity > LimitRTTIME=7000000 > IOSchedulingClass=realtime > IOSchedulingPriority=2 > CPUSchedulingPolicy=rr > CPUSchedulingPriority=89 > UMask=0007 > > [Install] > > > > > > > > This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited. > > > > Thank you for your compliance. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 > OpenTelecom.IT > cell: +39 347 266 56 18 > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at 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 at 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 at 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 > OpenTelecom.IT > cell: +39 347 266 56 18 > > This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited. > > > Thank you for your compliance. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From nbhatti at gmail.com Wed Jun 28 18:53:27 2017 From: nbhatti at gmail.com (Muhammad Naseer Bhatti) Date: Wed, 28 Jun 2017 14:53:27 -0400 Subject: [Freeswitch-users] How to delay setInputCallback() while playing audio? In-Reply-To: <86EB8621-BBF2-49F8-83BC-A139007D9E32@jerris.com> References: <86EB8621-BBF2-49F8-83BC-A139007D9E32@jerris.com> Message-ID: What about playback or streamFile from/to an offset and once this stream is finished, start another stream from where the last finished and then listen for events? session:streamFile(audioFile, 5000, 10000) and then session:streamFile(audioFile, 10001) and later listen for inputcallback. Offset is defined in playback and streamFile but no end point for the audio, so I guess that would not be possible? Or if there is a better logic to do that? -- Sent with Airmail From: Michael Jerris Reply: FreeSWITCH Users Help Date: June 28, 2017 at 11:17:42 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to delay setInputCallback() while playing audio? you’d have to handle ignoring for a couple seconds in your callback.. there is no way to delay setting the callback that I can think of. On Jun 28, 2017, at 2:02 PM, Muhammad Naseer Bhatti wrote: I guess this would only introduce a silence of 2 seconds. I am trying to play the file but delay DTMF detection for a few seconds. So my file is being played but I want to ignore a few seconds of DTMF inputs for callback function. Can’t think of a better way to do that. -- Sent with Airmail From: Brian West Reply: Brian West Date: June 28, 2017 at 10:12:00 PM To: Muhammad Naseer Bhatti Cc: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] How to delay setInputCallback() while playing audio? use streamFile silence_stream://2000 On Wed, Jun 28, 2017 at 11:48 AM, Muhammad Naseer Bhatti wrote: > > Or something like sched_api_function() which calls the function to detect > events? > > -- > > Sent with Airmail > > From: Muhammad Naseer Bhatti > Reply: Muhammad Naseer Bhatti > Date: June 28, 2017 at 12:18:40 AM > To: FreeSWITCH Users Help > , Brian West > > > Subject: Re: [Freeswitch-users] How to delay setInputCallback() while > playing audio? > > > playbackMessage() function is actually doing session:streamFile(audio_file). > Input callback is functioning fine, but I want to delay detection a few > seconds somehow. > > On the other hand, is there a way to stop a running session:streamFile() ? > > -- > > Sent with Airmail > > From: Brian West > Reply: FreeSWITCH Users Help > > Date: June 28, 2017 at 12:15:20 AM > To: FreeSWITCH Users Help > > Subject: Re: [Freeswitch-users] How to delay setInputCallback() while > playing audio? > > Show us how you're doing the playback, If you're doing > session:execute('playback', 'blah.wav'); The input callback won't function > you have to use session:streamFile instead. > > /b > > > On Tue, Jun 27, 2017 at 1:10 PM, Muhammad Naseer Bhatti > wrote: > >> I am running an IVR which collects DTMF input while the file is being >> played, written in Lua. I set InputCallback and then listen for DTMF >> events. If event found, take actions etc. >> >> session:setInputCallback("eventHandler") >> playbackMessage (playback_delay) >> >> Facing a strange issues with Verizon where they send a quick RTP Event >> write after starting the audio, which triggers the DTMF event and bad >> things happen in my script. This only happens within first 1 to 2 seconds >> of RTP. How can I have my InputCallback listen for events only after a few >> seconds while still playing the audio? or do I have to switch to >> playAndGetDigits() ? >> >> >> -- >> > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From rick at magicmail.mooo.com Wed Jun 28 18:59:15 2017 From: rick at magicmail.mooo.com (Rick Jarvis) Date: Wed, 28 Jun 2017 19:59:15 +0100 Subject: [Freeswitch-users] FS service issues (systemd) In-Reply-To: <1A108FA3-C445-48C4-A3AA-98D4B4A2B275@magicmail.mooo.com> References: <6B172391-E75C-4CA2-A2DE-2C0A652C19EC@magicmail.mooo.com> <518D9916-4387-4C50-9932-DE812A12856B@magicmail.mooo.com> <4773C447-BBDD-46BC-8584-4A776A726FF5@gmail.com> <828993D4-5964-4EE7-87B6-D15830E045D3@magicmail.mooo.com> <1A108FA3-C445-48C4-A3AA-98D4B4A2B275@magicmail.mooo.com> Message-ID: Ok so I’ve been using: deb http://files.freeswitch.org/repo/deb/debian/ jessie main But it seems I should have been using deb http://files.freeswitch.org/repo/deb/freeswitch-1.6/ jessie main That’s confusing. Should the old one still be there? > On 28 Jun 2017, at 19:52, Rick Jarvis wrote: > > Hmm, wondering why the deb repo is still dishing out 1.4? :( > >> On 28 Jun 2017, at 16:11, Mundkowsky, Robert > wrote: >> >> Few more suggestions: <> >> >> 1 Make sure run and log directories are owned by same user that you run it as. >> 2 The newer version of FS are more stable for us. >> >> Robert Mundkowsky >> >> From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org ] On Behalf Of Giovanni Maruzzelli >> Sent: Wednesday, June 28, 2017 10:58 AM >> To: FreeSWITCH Users Help > >> Subject: Re: [Freeswitch-users] FS service issues (systemd) >> >> >> >> On 28 June 2017 at 16:54, Rick Jarvis > wrote: >> Debian Jessie, FreeSWITCH Version 1.4.26-37~64bit (-37 64bit) >> >> It starts fine manually, and works if I restart it with systemctl restart, which is why I’m confused, but leaning towards it being a dependency not having started in time when it boots. I can probably fix it by reinstalling and/or upgrading, I’d just like to know why it’s failing ideally... >> >> Use latest stable: 1.6.18, at this moment, and reinstall fron scratch, on a fresh new Debian Jessie install >> >> -giovanni >> >> >> >> >> On 28 Jun 2017, at 15:44, Giovanni Maruzzelli > wrote: >> >> What exact platform are you using? Os/distro? FS version? Any virtualization? >> >> Has ever started correctly? >> >> Have you started it from command line (maybe as root user) and then the problems started to appear? >> >> Have you tried to reinstall from scratch following strictly the confluence instructions? >> >> Maybe file/directories permission problem? >> >> -giovanni >> >> On 28 June 2017 at 16:32, Rick Jarvis > wrote: >> Just looping with failed registrations: >> >> 2017-06-28 15:29:42.738155 [NOTICE] sofia_reg.c:448 Registering connection1 >> 2017-06-28 15:29:42.738155 [NOTICE] sofia_reg.c:448 Registering connection2 >> 2017-06-28 15:29:42.738155 [NOTICE] sofia_reg.c:448 Registering xprov >> 2017-06-28 15:29:42.738155 [ERR] sofia_reg.c:2392 connection2 Failed Registration with status Service Unavailable [503]. failure #101 >> 2017-06-28 15:29:42.738155 [ERR] sofia_reg.c:2392 xprov Failed Registration with status Service Unavailable [503]. failure #101 >> 2017-06-28 15:29:42.838117 [ERR] sofia_reg.c:2392 connection1 Failed Registration with status Service Unavailable [503]. failure #101 >> 2017-06-28 15:29:43.738108 [WARNING] sofia_reg.c:505 connection1 Failed Registration [503], setting retry to 30 seconds. >> 2017-06-28 15:29:43.738108 [WARNING] sofia_reg.c:505 connection2 Failed Registration [503], setting retry to 30 seconds. >> 2017-06-28 15:29:43.738108 [WARNING] sofia_reg.c:505 xprov Failed Registration [503], setting retry to 30 seconds. >> >> On 28 Jun 2017, at 14:52, Luis Jimenez > wrote: >> >> What about the freeswitch.log file when fs is hung? >> >> >> On Jun 28, 2017, at 09:42, Rick Jarvis > wrote: >> >> I’ve tried deleting the db files but no difference unfortunately. Starting manually or even just restarting the service works, so I believe the XML is all as it should be. This is my output from systemctl when it’s in this ‘hung' state: >> >> ● freeswitch.service - freeswitch >> Loaded: loaded (/etc/systemd/system/freeswitch.service; enabled) >> Active: active (running) since Wed 2017-06-28 14:36:23 BST; 3min 27s ago >> Process: 452 ExecStart=/usr/bin/freeswitch -ncwait -nonat (code=exited, status=0/SUCCESS) >> Main PID: 472 (freeswitch) >> CGroup: /system.slice/freeswitch.service >> └─472 /usr/bin/freeswitch -ncwait -nonat >> >> Jun 28 14:36:20 server systemd[1]: Starting freeswitch... >> Jun 28 14:36:20 server freeswitch[452]: 472 Backgrounding. >> Jun 28 14:36:23 server freeswitch[452]: FreeSWITCH[452] Waiting for background process pid:472 to be ready..... >> Jun 28 14:36:23 server freeswitch[452]: FreeSWITCH[452] System Ready pid:472 >> Jun 28 14:36:23 server systemd[1]: Started freeswitch. >> >> >> On 28 Jun 2017, at 14:18, Mundkowsky, Robert > wrote: >> >> Yes. I think we had a similar problem in ,I think, two situations. <> >> >> 1) The file system database can get messed up. >> a. Stop FS >> b. You can clear then via rm /export/Apps/freeswitch/var/lib/freeswitch/db/* >> c. Restart FS >> 2) If there is an error in the configuration files. >> a. Take a look at /export/Apps/freeswitch/var/log/freeswitch/freeswitch.xml.fsxml to see if there are any errors in XML. >> >> You can get more details of start problem via: >> journalctl -xn >> systemctl status freeswitch.service >> >> if you do not get enough details, try starting FS directly and you might get more details via: >> >> /…/bin/freeswitch >> >> >> Robert Mundkowsky >> >> From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org ] On Behalf Of Rick Jarvis >> Sent: Wednesday, June 28, 2017 7:44 AM >> To: FreeSWITCH Users Help > >> Subject: [Freeswitch-users] FS service issues (systemd) >> >> >> Got a weird issue where FS isn’t starting up correctly. Looking at the logs, it appears just to be attempting to restart SIP profiles (and failing), and that’s about it. There’s no socket for fs_cli, and a simple ‘systemctl restart’ fixes it. Anyone had something like this? I’ve tried fixing it with systemd after / requires, but it hasn’t helped. >> >> Here’s my .service file (not sure if this is the issue, I'd welcome any comments / better versions?) >> >> [Unit] >> Description=freeswitch >> After=syslog.target network-online.target local-fs.target >> >> [Service] >> ; service >> Type=forking >> PIDFile=/run/freeswitch/freeswitch.pid >> PermissionsStartOnly=true >> ExecStart=/usr/bin/freeswitch -ncwait -nonat >> ExecStop=/usr/bin/fs_cli -x shutdown >> TimeoutSec=45s >> Restart=always >> ; exec >> RuntimeDirectory=freeswitch >> RuntimeDirectoryMode=0755 >> User=freeswitch >> Group=freeswitch >> LimitCORE=infinity >> LimitNOFILE=100000 >> LimitNPROC=60000 >> ;LimitSTACK=240 >> LimitRTPRIO=infinity >> LimitRTTIME=7000000 >> IOSchedulingClass=realtime >> IOSchedulingPriority=2 >> CPUSchedulingPolicy=rr >> CPUSchedulingPriority=89 >> UMask=0007 >> >> [Install] >> >> >> >> >> >> >> >> This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited. >> >> >> >> Thank you for your compliance. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 >> OpenTelecom.IT >> cell: +39 347 266 56 18 >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at 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 at 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 at 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 >> OpenTelecom.IT >> cell: +39 347 266 56 18 >> >> This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited. >> >> >> Thank you for your compliance. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From gmaruzz at gmail.com Wed Jun 28 19:08:01 2017 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Wed, 28 Jun 2017 21:08:01 +0200 Subject: [Freeswitch-users] FS service issues (systemd) In-Reply-To: References: <6B172391-E75C-4CA2-A2DE-2C0A652C19EC@magicmail.mooo.com> <518D9916-4387-4C50-9932-DE812A12856B@magicmail.mooo.com> <4773C447-BBDD-46BC-8584-4A776A726FF5@gmail.com> <828993D4-5964-4EE7-87B6-D15830E045D3@magicmail.mooo.com> <1A108FA3-C445-48C4-A3AA-98D4B4A2B275@magicmail.mooo.com> Message-ID: On 28 June 2017 at 20:59, Rick Jarvis wrote: > Ok so I’ve been using: > > deb http://files.freeswitch.org/repo/deb/debian/ jessie main > > But it seems I should have been using > > deb http://files.freeswitch.org/repo/deb/freeswitch-1.6/ jessie main > > That’s confusing. Should the old one still be there? > > Best is to follow the instruction, until you know better ;) https://freeswitch.org/confluence/display/FREESWITCH/Debian+8+Jessie -giovanni -------------- next part -------------- An HTML attachment was scrubbed... URL: From rick at magicmail.mooo.com Wed Jun 28 19:14:54 2017 From: rick at magicmail.mooo.com (Rick Jarvis) Date: Wed, 28 Jun 2017 20:14:54 +0100 Subject: [Freeswitch-users] FS service issues (systemd) In-Reply-To: References: <6B172391-E75C-4CA2-A2DE-2C0A652C19EC@magicmail.mooo.com> <518D9916-4387-4C50-9932-DE812A12856B@magicmail.mooo.com> <4773C447-BBDD-46BC-8584-4A776A726FF5@gmail.com> <828993D4-5964-4EE7-87B6-D15830E045D3@magicmail.mooo.com> <1A108FA3-C445-48C4-A3AA-98D4B4A2B275@magicmail.mooo.com> Message-ID: <575D7EAA-6BDB-4B5C-BEB9-2A2E7781BBE1@magicmail.mooo.com> I thought I did know better ;) Actually I’m not sure where I got the first repo url from, but I think I always assumed that as there was no version number in it, that it would be the latest stable release. Is it worth taking the non-versioned one down, just a thought? ;) > On 28 Jun 2017, at 20:08, Giovanni Maruzzelli wrote: > > On 28 June 2017 at 20:59, Rick Jarvis > wrote: > Ok so I’ve been using: > > deb http://files.freeswitch.org/repo/deb/debian/ jessie main > > But it seems I should have been using > > deb http://files.freeswitch.org/repo/deb/freeswitch-1.6/ jessie main > > That’s confusing. Should the old one still be there? > > > > Best is to follow the instruction, until you know better ;) > > https://freeswitch.org/confluence/display/FREESWITCH/Debian+8+Jessie > > -giovanni > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From krice at freeswitch.org Wed Jun 28 19:19:34 2017 From: krice at freeswitch.org (Ken Rice) Date: Wed, 28 Jun 2017 14:19:34 -0500 Subject: [Freeswitch-users] FS service issues (systemd) In-Reply-To: <575D7EAA-6BDB-4B5C-BEB9-2A2E7781BBE1@magicmail.mooo.com> References: <6B172391-E75C-4CA2-A2DE-2C0A652C19EC@magicmail.mooo.com> <518D9916-4387-4C50-9932-DE812A12856B@magicmail.mooo.com> <4773C447-BBDD-46BC-8584-4A776A726FF5@gmail.com> <828993D4-5964-4EE7-87B6-D15830E045D3@magicmail.mooo.com> <1A108FA3-C445-48C4-A3AA-98D4B4A2B275@magicmail.mooo.com> <575D7EAA-6BDB-4B5C-BEB9-2A2E7781BBE1@magicmail.mooo.com> Message-ID: <0a2601d2f043$7a7c7ea0$6f757be0$@freeswitch.org> Cant really take the non-versioned one down… that’s the original URL and theres still tons of things pointing at it. We do our best to keep installation instructions up to date, and put notices on things that are deprecated, but people still keep using them. There’s even a popup on the old wiki that says “hey go to confluence this is old and will not be updated ever again”… they still contact me wanting to know the info on the wiki is wrong. K From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rick Jarvis Sent: Wednesday, June 28, 2017 2:15 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] FS service issues (systemd) I thought I did know better ;) Actually I’m not sure where I got the first repo url from, but I think I always assumed that as there was no version number in it, that it would be the latest stable release. Is it worth taking the non-versioned one down, just a thought? ;) On 28 Jun 2017, at 20:08, Giovanni Maruzzelli > wrote: On 28 June 2017 at 20:59, Rick Jarvis > wrote: Ok so I’ve been using: deb http://files.freeswitch.org/repo/deb/debian/ jessie main But it seems I should have been using deb http://files.freeswitch.org/repo/deb/freeswitch-1.6/ jessie main That’s confusing. Should the old one still be there? Best is to follow the instruction, until you know better ;) https://freeswitch.org/confluence/display/FREESWITCH/Debian+8+Jessie -giovanni _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From rick at magicmail.mooo.com Wed Jun 28 19:25:05 2017 From: rick at magicmail.mooo.com (Rick Jarvis) Date: Wed, 28 Jun 2017 20:25:05 +0100 Subject: [Freeswitch-users] FS service issues (systemd) In-Reply-To: <0a2601d2f043$7a7c7ea0$6f757be0$@freeswitch.org> References: <6B172391-E75C-4CA2-A2DE-2C0A652C19EC@magicmail.mooo.com> <518D9916-4387-4C50-9932-DE812A12856B@magicmail.mooo.com> <4773C447-BBDD-46BC-8584-4A776A726FF5@gmail.com> <828993D4-5964-4EE7-87B6-D15830E045D3@magicmail.mooo.com> <1A108FA3-C445-48C4-A3AA-98D4B4A2B275@magicmail.mooo.com> <575D7EAA-6BDB-4B5C-BEB9-2A2E7781BBE1@magicmail.mooo.com> <0a2601d2f043$7a7c7ea0$6f757be0$@freeswitch.org> Message-ID: <28ABCDDD-27A7-4AED-8B2E-633D89865177@magicmail.mooo.com> Fair enough Ken. I guess you can’t always account for stoopid ;) > On 28 Jun 2017, at 20:19, Ken Rice wrote: > > Cant really take the non-versioned one down… that’s the original URL and theres still tons of things pointing at it. We do our best to keep installation instructions up to date, and put notices on things that are deprecated, but people still keep using them. > > There’s even a popup on the old wiki that says “hey go to confluence this is old and will not be updated ever again”… they still contact me wanting to know the info on the wiki is wrong. > > K > > From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org ] On Behalf Of Rick Jarvis > Sent: Wednesday, June 28, 2017 2:15 PM > To: FreeSWITCH Users Help > > Subject: Re: [Freeswitch-users] FS service issues (systemd) > > I thought I did know better ;) > > Actually I’m not sure where I got the first repo url from, but I think I always assumed that as there was no version number in it, that it would be the latest stable release. Is it worth taking the non-versioned one down, just a thought? ;) > >> On 28 Jun 2017, at 20:08, Giovanni Maruzzelli > wrote: >> >> On 28 June 2017 at 20:59, Rick Jarvis > wrote: >>> Ok so I’ve been using: >>> >>> deb http://files.freeswitch.org/repo/deb/debian/ jessie main >>> >>> But it seems I should have been using >>> >>> deb http://files.freeswitch.org/repo/deb/freeswitch-1.6/ jessie main >>> >>> That’s confusing. Should the old one still be there? >>> >> >> >> Best is to follow the instruction, until you know better ;) >> >> https://freeswitch.org/confluence/display/FREESWITCH/Debian+8+Jessie >> -giovanni >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From jkomar at jbox.ca Wed Jun 28 19:26:36 2017 From: jkomar at jbox.ca (Jason Komar) Date: Wed, 28 Jun 2017 13:26:36 -0600 Subject: [Freeswitch-users] FS service issues (systemd) In-Reply-To: References: <6B172391-E75C-4CA2-A2DE-2C0A652C19EC@magicmail.mooo.com> <518D9916-4387-4C50-9932-DE812A12856B@magicmail.mooo.com> <4773C447-BBDD-46BC-8584-4A776A726FF5@gmail.com> <828993D4-5964-4EE7-87B6-D15830E045D3@magicmail.mooo.com> Message-ID: I had this happen on one particular FS box and no matter what I tried I could not get FS to wait until the NIC was 100% up and running. I ended up installing a different model NIC and it started working. I can't remember what model the troublesome one was, unfortunately. Jason On Wed, Jun 28, 2017 at 8:54 AM, Rick Jarvis wrote: > Debian Jessie, FreeSWITCH Version 1.4.26-37~64bit (-37 64bit) > > It starts fine manually, and works if I restart it with systemctl restart, > which is why I’m confused, but leaning towards it being a dependency not > having started in time when it boots. I can probably fix it by reinstalling > and/or upgrading, I’d just like to know why it’s failing ideally... > > > On 28 Jun 2017, at 15:44, Giovanni Maruzzelli wrote: > > What exact platform are you using? Os/distro? FS version? Any > virtualization? > > Has ever started correctly? > > Have you started it from command line (maybe as root user) and then the > problems started to appear? > > Have you tried to reinstall from scratch following strictly the confluence > instructions? > > Maybe file/directories permission problem? > > -giovanni > > On 28 June 2017 at 16:32, Rick Jarvis wrote: > >> Just looping with failed registrations: >> >> 2017-06-28 15:29:42.738155 [NOTICE] sofia_reg.c:448 Registering >> connection1 >> 2017-06-28 15:29:42.738155 [NOTICE] sofia_reg.c:448 Registering >> connection2 >> 2017-06-28 15:29:42.738155 [NOTICE] sofia_reg.c:448 Registering xprov >> 2017-06-28 15:29:42.738155 [ERR] sofia_reg.c:2392 connection2 Failed >> Registration with status Service Unavailable [503]. failure #101 >> 2017-06-28 15:29:42.738155 [ERR] sofia_reg.c:2392 xprov Failed >> Registration with status Service Unavailable [503]. failure #101 >> 2017-06-28 15:29:42.838117 [ERR] sofia_reg.c:2392 connection1 Failed >> Registration with status Service Unavailable [503]. failure #101 >> 2017-06-28 15:29:43.738108 [WARNING] sofia_reg.c:505 connection1 Failed >> Registration [503], setting retry to 30 seconds. >> 2017-06-28 15:29:43.738108 [WARNING] sofia_reg.c:505 connection2 Failed >> Registration [503], setting retry to 30 seconds. >> 2017-06-28 15:29:43.738108 [WARNING] sofia_reg.c:505 xprov Failed >> Registration [503], setting retry to 30 seconds. >> >> On 28 Jun 2017, at 14:52, Luis Jimenez wrote: >> >> What about the freeswitch.log file when fs is hung? >> >> >> On Jun 28, 2017, at 09:42, Rick Jarvis wrote: >> >> I’ve tried deleting the db files but no difference unfortunately. >> Starting manually or even just restarting the service works, so I believe >> the XML is all as it should be. This is my output from systemctl when it’s >> in this ‘hung' state: >> >> ● freeswitch.service - freeswitch >> Loaded: loaded (/etc/systemd/system/freeswitch.service; enabled) >> Active: active (running) since Wed 2017-06-28 14:36:23 BST; 3min 27s >> ago >> Process: 452 ExecStart=/usr/bin/freeswitch -ncwait -nonat (code=exited, >> status=0/SUCCESS) >> Main PID: 472 (freeswitch) >> CGroup: /system.slice/freeswitch.service >> └─472 /usr/bin/freeswitch -ncwait -nonat >> >> Jun 28 14:36:20 server systemd[1]: Starting freeswitch... >> Jun 28 14:36:20 server freeswitch[452]: 472 Backgrounding. >> Jun 28 14:36:23 server freeswitch[452]: FreeSWITCH[452] Waiting for >> background process pid:472 to be ready..... >> Jun 28 14:36:23 server freeswitch[452]: FreeSWITCH[452] System Ready >> pid:472 >> Jun 28 14:36:23 server systemd[1]: Started freeswitch. >> >> >> On 28 Jun 2017, at 14:18, Mundkowsky, Robert wrote: >> >> Yes. I think we had a similar problem in ,I think, two situations. >> >> 1) The file system database can get messed up. >> a. Stop FS >> b. You can clear then via rm /export/Apps/freeswitch/var/li >> b/freeswitch/db/* >> c. Restart FS >> 2) If there is an error in the configuration files. >> a. Take a look at /export/Apps/freeswitch/var/lo >> g/freeswitch/freeswitch.xml.fsxml to see if there are any errors in XML. >> >> You can get more details of start problem via: >> journalctl -xn >> systemctl status freeswitch.service >> >> if you do not get enough details, try starting FS >> directly and you might get more details via: >> >> /…/bin/freeswitch >> >> >> Robert Mundkowsky >> >> *From:* FreeSWITCH-users [mailto:freeswitch-users-bounc >> es at lists.freeswitch.org ] *On >> Behalf Of *Rick Jarvis >> *Sent:* Wednesday, June 28, 2017 7:44 AM >> *To:* FreeSWITCH Users Help >> *Subject:* [Freeswitch-users] FS service issues (systemd) >> >> >> Got a weird issue where FS isn’t starting up correctly. Looking at the >> logs, it appears just to be attempting to restart SIP profiles (and >> failing), and that’s about it. There’s no socket for fs_cli, and a simple >> ‘systemctl restart’ fixes it. Anyone had something like this? I’ve tried >> fixing it with systemd after / requires, but it hasn’t helped. >> >> Here’s my .service file (not sure if this is the issue, I'd welcome any >> comments / better versions?) >> >> [Unit] >> Description=freeswitch >> After=syslog.target network-online.target local-fs.target >> >> [Service] >> ; service >> Type=forking >> PIDFile=/run/freeswitch/freeswitch.pid >> PermissionsStartOnly=true >> ExecStart=/usr/bin/freeswitch -ncwait -nonat >> ExecStop=/usr/bin/fs_cli -x shutdown >> TimeoutSec=45s >> Restart=always >> ; exec >> RuntimeDirectory=freeswitch >> RuntimeDirectoryMode=0755 >> User=freeswitch >> Group=freeswitch >> LimitCORE=infinity >> LimitNOFILE=100000 >> LimitNPROC=60000 >> ;LimitSTACK=240 >> LimitRTPRIO=infinity >> LimitRTTIME=7000000 >> IOSchedulingClass=realtime >> IOSchedulingPriority=2 >> CPUSchedulingPolicy=rr >> CPUSchedulingPriority=89 >> UMask=0007 >> >> [Install] >> >> >> >> >> >> >> >> ------------------------------ >> >> This e-mail and any files transmitted with it may contain privileged or >> confidential information. It is solely for use by the individual for whom >> it is intended, even if addressed incorrectly. If you received this e-mail >> in error, please notify the sender; do not disclose, copy, distribute, or >> take any action in reliance on the contents of this information; and delete >> it from your system. Any other use of this e-mail is prohibited. >> >> Thank you for your compliance. >> ------------------------------ >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 > OpenTelecom.IT > cell: +39 347 266 56 18 > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From rick at magicmail.mooo.com Wed Jun 28 19:48:45 2017 From: rick at magicmail.mooo.com (Rick Jarvis) Date: Wed, 28 Jun 2017 20:48:45 +0100 Subject: [Freeswitch-users] FS service issues (systemd) In-Reply-To: References: <6B172391-E75C-4CA2-A2DE-2C0A652C19EC@magicmail.mooo.com> <518D9916-4387-4C50-9932-DE812A12856B@magicmail.mooo.com> <4773C447-BBDD-46BC-8584-4A776A726FF5@gmail.com> <828993D4-5964-4EE7-87B6-D15830E045D3@magicmail.mooo.com> Message-ID: Thanks Jason. Indeed, fresh install of deb 8.8 and FS 1.6, and exactly the same behaviour. Needs a manual ‘systemctl restart freeswitch’ to get it into life. Quite annoying, especially as it was ok before… not quite sure what’s changed. > On 28 Jun 2017, at 20:26, Jason Komar wrote: > > I had this happen on one particular FS box and no matter what I tried I could not get FS to wait until the NIC was 100% up and running. I ended up installing a different model NIC and it started working. I can't remember what model the troublesome one was, unfortunately. > > Jason > > > On Wed, Jun 28, 2017 at 8:54 AM, Rick Jarvis > wrote: > Debian Jessie, FreeSWITCH Version 1.4.26-37~64bit (-37 64bit) > > It starts fine manually, and works if I restart it with systemctl restart, which is why I’m confused, but leaning towards it being a dependency not having started in time when it boots. I can probably fix it by reinstalling and/or upgrading, I’d just like to know why it’s failing ideally... > > >> On 28 Jun 2017, at 15:44, Giovanni Maruzzelli > wrote: >> >> What exact platform are you using? Os/distro? FS version? Any virtualization? >> >> Has ever started correctly? >> >> Have you started it from command line (maybe as root user) and then the problems started to appear? >> >> Have you tried to reinstall from scratch following strictly the confluence instructions? >> >> Maybe file/directories permission problem? >> >> -giovanni >> >> On 28 June 2017 at 16:32, Rick Jarvis > wrote: >> Just looping with failed registrations: >> >> 2017-06-28 15:29:42.738155 [NOTICE] sofia_reg.c:448 Registering connection1 >> 2017-06-28 15:29:42.738155 [NOTICE] sofia_reg.c:448 Registering connection2 >> 2017-06-28 15:29:42.738155 [NOTICE] sofia_reg.c:448 Registering xprov >> 2017-06-28 15:29:42.738155 [ERR] sofia_reg.c:2392 connection2 Failed Registration with status Service Unavailable [503]. failure #101 >> 2017-06-28 15:29:42.738155 [ERR] sofia_reg.c:2392 xprov Failed Registration with status Service Unavailable [503]. failure #101 >> 2017-06-28 15:29:42.838117 [ERR] sofia_reg.c:2392 connection1 Failed Registration with status Service Unavailable [503]. failure #101 >> 2017-06-28 15:29:43.738108 [WARNING] sofia_reg.c:505 connection1 Failed Registration [503], setting retry to 30 seconds. >> 2017-06-28 15:29:43.738108 [WARNING] sofia_reg.c:505 connection2 Failed Registration [503], setting retry to 30 seconds. >> 2017-06-28 15:29:43.738108 [WARNING] sofia_reg.c:505 xprov Failed Registration [503], setting retry to 30 seconds. >> >>> On 28 Jun 2017, at 14:52, Luis Jimenez > wrote: >>> >>> What about the freeswitch.log file when fs is hung? >>> >>> >>> On Jun 28, 2017, at 09:42, Rick Jarvis > wrote: >>> >>>> I’ve tried deleting the db files but no difference unfortunately. Starting manually or even just restarting the service works, so I believe the XML is all as it should be. This is my output from systemctl when it’s in this ‘hung' state: >>>> >>>> ● freeswitch.service - freeswitch >>>> Loaded: loaded (/etc/systemd/system/freeswitch.service; enabled) >>>> Active: active (running) since Wed 2017-06-28 14:36:23 BST; 3min 27s ago >>>> Process: 452 ExecStart=/usr/bin/freeswitch -ncwait -nonat (code=exited, status=0/SUCCESS) >>>> Main PID: 472 (freeswitch) >>>> CGroup: /system.slice/freeswitch.service >>>> └─472 /usr/bin/freeswitch -ncwait -nonat >>>> >>>> Jun 28 14:36:20 server systemd[1]: Starting freeswitch... >>>> Jun 28 14:36:20 server freeswitch[452]: 472 Backgrounding. >>>> Jun 28 14:36:23 server freeswitch[452]: FreeSWITCH[452] Waiting for background process pid:472 to be ready..... >>>> Jun 28 14:36:23 server freeswitch[452]: FreeSWITCH[452] System Ready pid:472 >>>> Jun 28 14:36:23 server systemd[1]: Started freeswitch. >>>> >>>> >>>>> On 28 Jun 2017, at 14:18, Mundkowsky, Robert > wrote: >>>>> >>>>> Yes. I think we had a similar problem in ,I think, two situations. <> >>>>> >>>>> 1) The file system database can get messed up. >>>>> a. Stop FS >>>>> b. You can clear then via rm /export/Apps/freeswitch/var/lib/freeswitch/db/* >>>>> c. Restart FS >>>>> 2) If there is an error in the configuration files. >>>>> a. Take a look at /export/Apps/freeswitch/var/log/freeswitch/freeswitch.xml.fsxml to see if there are any errors in XML. >>>>> >>>>> You can get more details of start problem via: >>>>> journalctl -xn >>>>> systemctl status freeswitch.service >>>>> >>>>> if you do not get enough details, try starting FS directly and you might get more details via: >>>>> >>>>> /…/bin/freeswitch >>>>> >>>>> >>>>> Robert Mundkowsky >>>>> >>>>> From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org ] On Behalf Of Rick Jarvis >>>>> Sent: Wednesday, June 28, 2017 7:44 AM >>>>> To: FreeSWITCH Users Help > >>>>> Subject: [Freeswitch-users] FS service issues (systemd) >>>>> >>>>> >>>>> Got a weird issue where FS isn’t starting up correctly. Looking at the logs, it appears just to be attempting to restart SIP profiles (and failing), and that’s about it. There’s no socket for fs_cli, and a simple ‘systemctl restart’ fixes it. Anyone had something like this? I’ve tried fixing it with systemd after / requires, but it hasn’t helped. >>>>> >>>>> Here’s my .service file (not sure if this is the issue, I'd welcome any comments / better versions?) >>>>> >>>>> [Unit] >>>>> Description=freeswitch >>>>> After=syslog.target network-online.target local-fs.target >>>>> >>>>> [Service] >>>>> ; service >>>>> Type=forking >>>>> PIDFile=/run/freeswitch/freeswitch.pid >>>>> PermissionsStartOnly=true >>>>> ExecStart=/usr/bin/freeswitch -ncwait -nonat >>>>> ExecStop=/usr/bin/fs_cli -x shutdown >>>>> TimeoutSec=45s >>>>> Restart=always >>>>> ; exec >>>>> RuntimeDirectory=freeswitch >>>>> RuntimeDirectoryMode=0755 >>>>> User=freeswitch >>>>> Group=freeswitch >>>>> LimitCORE=infinity >>>>> LimitNOFILE=100000 >>>>> LimitNPROC=60000 >>>>> ;LimitSTACK=240 >>>>> LimitRTPRIO=infinity >>>>> LimitRTTIME=7000000 >>>>> IOSchedulingClass=realtime >>>>> IOSchedulingPriority=2 >>>>> CPUSchedulingPolicy=rr >>>>> CPUSchedulingPriority=89 >>>>> UMask=0007 >>>>> >>>>> [Install] >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited. >>>>> >>>>> >>>>> Thank you for your compliance. >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 >> OpenTelecom.IT >> cell: +39 347 266 56 18 >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From krice at freeswitch.org Wed Jun 28 19:51:47 2017 From: krice at freeswitch.org (Ken Rice) Date: Wed, 28 Jun 2017 14:51:47 -0500 Subject: [Freeswitch-users] FS service issues (systemd) In-Reply-To: References: <6B172391-E75C-4CA2-A2DE-2C0A652C19EC@magicmail.mooo.com> <518D9916-4387-4C50-9932-DE812A12856B@magicmail.mooo.com> <4773C447-BBDD-46BC-8584-4A776A726FF5@gmail.com> <828993D4-5964-4EE7-87B6-D15830E045D3@magicmail.mooo.com> Message-ID: <0a5201d2f047$fa8f8660$efae9320$@freeswitch.org> Try changing it so that FreeSWITCH depends on the network to be full up before it’ll start in the system setup. Theres a way to do that but the data eludes me at this time From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rick Jarvis Sent: Wednesday, June 28, 2017 2:49 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] FS service issues (systemd) Thanks Jason. Indeed, fresh install of deb 8.8 and FS 1.6, and exactly the same behaviour. Needs a manual ‘systemctl restart freeswitch’ to get it into life. Quite annoying, especially as it was ok before… not quite sure what’s changed. On 28 Jun 2017, at 20:26, Jason Komar > wrote: I had this happen on one particular FS box and no matter what I tried I could not get FS to wait until the NIC was 100% up and running. I ended up installing a different model NIC and it started working. I can't remember what model the troublesome one was, unfortunately. Jason On Wed, Jun 28, 2017 at 8:54 AM, Rick Jarvis > wrote: Debian Jessie, FreeSWITCH Version 1.4.26-37~64bit (-37 64bit) It starts fine manually, and works if I restart it with systemctl restart, which is why I’m confused, but leaning towards it being a dependency not having started in time when it boots. I can probably fix it by reinstalling and/or upgrading, I’d just like to know why it’s failing ideally... On 28 Jun 2017, at 15:44, Giovanni Maruzzelli > wrote: What exact platform are you using? Os/distro? FS version? Any virtualization? Has ever started correctly? Have you started it from command line (maybe as root user) and then the problems started to appear? Have you tried to reinstall from scratch following strictly the confluence instructions? Maybe file/directories permission problem? -giovanni On 28 June 2017 at 16:32, Rick Jarvis > wrote: Just looping with failed registrations: 2017-06-28 15:29:42.738155 [NOTICE] sofia_reg.c:448 Registering connection1 2017-06-28 15:29:42.738155 [NOTICE] sofia_reg.c:448 Registering connection2 2017-06-28 15:29:42.738155 [NOTICE] sofia_reg.c:448 Registering xprov 2017-06-28 15:29:42.738155 [ERR] sofia_reg.c:2392 connection2 Failed Registration with status Service Unavailable [503]. failure #101 2017-06-28 15:29:42.738155 [ERR] sofia_reg.c:2392 xprov Failed Registration with status Service Unavailable [503]. failure #101 2017-06-28 15:29:42.838117 [ERR] sofia_reg.c:2392 connection1 Failed Registration with status Service Unavailable [503]. failure #101 2017-06-28 15:29:43.738108 [WARNING] sofia_reg.c:505 connection1 Failed Registration [503], setting retry to 30 seconds. 2017-06-28 15:29:43.738108 [WARNING] sofia_reg.c:505 connection2 Failed Registration [503], setting retry to 30 seconds. 2017-06-28 15:29:43.738108 [WARNING] sofia_reg.c:505 xprov Failed Registration [503], setting retry to 30 seconds. On 28 Jun 2017, at 14:52, Luis Jimenez > wrote: What about the freeswitch.log file when fs is hung? On Jun 28, 2017, at 09:42, Rick Jarvis > wrote: I’ve tried deleting the db files but no difference unfortunately. Starting manually or even just restarting the service works, so I believe the XML is all as it should be. This is my output from systemctl when it’s in this ‘hung' state: ● freeswitch.service - freeswitch Loaded: loaded (/etc/systemd/system/freeswitch.service; enabled) Active: active (running) since Wed 2017-06-28 14:36:23 BST; 3min 27s ago Process: 452 ExecStart=/usr/bin/freeswitch -ncwait -nonat (code=exited, status=0/SUCCESS) Main PID: 472 (freeswitch) CGroup: /system.slice/freeswitch.service └─472 /usr/bin/freeswitch -ncwait -nonat Jun 28 14:36:20 server systemd[1]: Starting freeswitch... Jun 28 14:36:20 server freeswitch[452]: 472 Backgrounding. Jun 28 14:36:23 server freeswitch[452]: FreeSWITCH[452] Waiting for background process pid:472 to be ready..... Jun 28 14:36:23 server freeswitch[452]: FreeSWITCH[452] System Ready pid:472 Jun 28 14:36:23 server systemd[1]: Started freeswitch. On 28 Jun 2017, at 14:18, Mundkowsky, Robert > wrote: Yes. I think we had a similar problem in ,I think, two situations. 1) The file system database can get messed up. a. Stop FS b. You can clear then via rm /export/Apps/freeswitch/var/lib/freeswitch/db/* c. Restart FS 2) If there is an error in the configuration files. a. Take a look at /export/Apps/freeswitch/var/log/freeswitch/freeswitch.xml.fsxml to see if there are any errors in XML. You can get more details of start problem via: journalctl -xn systemctl status freeswitch.service if you do not get enough details, try starting FS directly and you might get more details via: /…/bin/freeswitch Robert Mundkowsky From: FreeSWITCH-users [ mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rick Jarvis Sent: Wednesday, June 28, 2017 7:44 AM To: FreeSWITCH Users Help < freeswitch-users at lists.freeswitch.org> Subject: [Freeswitch-users] FS service issues (systemd) Got a weird issue where FS isn’t starting up correctly. Looking at the logs, it appears just to be attempting to restart SIP profiles (and failing), and that’s about it. There’s no socket for fs_cli, and a simple ‘systemctl restart’ fixes it. Anyone had something like this? I’ve tried fixing it with systemd after / requires, but it hasn’t helped. Here’s my .service file (not sure if this is the issue, I'd welcome any comments / better versions?) [Unit] Description=freeswitch After=syslog.target network-online.target local-fs.target [Service] ; service Type=forking PIDFile=/run/freeswitch/freeswitch.pid PermissionsStartOnly=true ExecStart=/usr/bin/freeswitch -ncwait -nonat ExecStop=/usr/bin/fs_cli -x shutdown TimeoutSec=45s Restart=always ; exec RuntimeDirectory=freeswitch RuntimeDirectoryMode=0755 User=freeswitch Group=freeswitch LimitCORE=infinity LimitNOFILE=100000 LimitNPROC=60000 ;LimitSTACK=240 LimitRTPRIO=infinity LimitRTTIME=7000000 IOSchedulingClass=realtime IOSchedulingPriority=2 CPUSchedulingPolicy=rr CPUSchedulingPriority=89 UMask=0007 [Install] _____ This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited. Thank you for your compliance. _____ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 OpenTelecom.IT cell: +39 347 266 56 18 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at 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 at 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 at 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 at 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 at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From infos at madovsky.org Wed Jun 28 19:56:43 2017 From: infos at madovsky.org (Madovsky) Date: Wed, 28 Jun 2017 12:56:43 -0700 Subject: [Freeswitch-users] FS service issues (systemd) In-Reply-To: References: <6B172391-E75C-4CA2-A2DE-2C0A652C19EC@magicmail.mooo.com> <518D9916-4387-4C50-9932-DE812A12856B@magicmail.mooo.com> <4773C447-BBDD-46BC-8584-4A776A726FF5@gmail.com> <828993D4-5964-4EE7-87B6-D15830E045D3@magicmail.mooo.com> Message-ID: systemd has always been a freaking hell since 4 years.... > Indeed, fresh install of deb 8.8 and FS 1.6, and exactly the same > behaviour. Needs a manual ‘systemctl restart freeswitch’ to get it > into life. Quite annoying, especially as it was ok before… not quite > sure what’s changed. > >> On 28 Jun 2017, at 20:26, Jason Komar > > wrote: >> >> I had this happen on one particular FS box and no matter what I tried >> I could not get FS to wait until the NIC was 100% up and running. I >> ended up installing a different model NIC and it started working. I >> can't remember what model the troublesome one was, unfortunately. >> >> Jason >> >> >> On Wed, Jun 28, 2017 at 8:54 AM, Rick Jarvis > > wrote: >> >> Debian Jessie, FreeSWITCH Version 1.4.26-37~64bit (-37 64bit) >> >> It starts fine manually, and works if I restart it with systemctl >> restart, which is why I’m confused, but leaning towards it being >> a dependency not having started in time when it boots. I can >> probably fix it by reinstalling and/or upgrading, I’d just like >> to know why it’s failing ideally... >> >> >>> On 28 Jun 2017, at 15:44, Giovanni Maruzzelli >> > wrote: >>> >>> What exact platform are you using? Os/distro? FS version? Any >>> virtualization? >>> >>> Has ever started correctly? >>> >>> Have you started it from command line (maybe as root user) and >>> then the problems started to appear? >>> >>> Have you tried to reinstall from scratch following strictly the >>> confluence instructions? >>> >>> Maybe file/directories permission problem? >>> >>> -giovanni >>> >>> On 28 June 2017 at 16:32, Rick Jarvis >> > wrote: >>> >>> Just looping with failed registrations: >>> >>> 2017-06-28 15:29:42.738155 [NOTICE] sofia_reg.c:448 >>> Registering connection1 >>> 2017-06-28 15:29:42.738155 [NOTICE] sofia_reg.c:448 >>> Registering connection2 >>> 2017-06-28 15:29:42.738155 [NOTICE] sofia_reg.c:448 >>> Registering xprov >>> 2017-06-28 15:29:42.738155 [ERR] sofia_reg.c:2392 >>> connection2 Failed Registration with status Service >>> Unavailable [503]. failure #101 >>> 2017-06-28 15:29:42.738155 [ERR] sofia_reg.c:2392 xprov >>> Failed Registration with status Service Unavailable [503]. >>> failure #101 >>> 2017-06-28 15:29:42.838117 [ERR] sofia_reg.c:2392 >>> connection1 Failed Registration with status Service >>> Unavailable [503]. failure #101 >>> 2017-06-28 15:29:43.738108 [WARNING] sofia_reg.c:505 >>> connection1 Failed Registration [503], setting retry to 30 >>> seconds. >>> 2017-06-28 15:29:43.738108 [WARNING] sofia_reg.c:505 >>> connection2 Failed Registration [503], setting retry to 30 >>> seconds. >>> 2017-06-28 15:29:43.738108 [WARNING] sofia_reg.c:505 xprov >>> Failed Registration [503], setting retry to 30 seconds. >>> >>>> On 28 Jun 2017, at 14:52, Luis Jimenez >>> > wrote: >>>> >>>> What about the freeswitch.log file when fs is hung? >>>> >>>> >>>> On Jun 28, 2017, at 09:42, Rick Jarvis >>>> > >>>> wrote: >>>> >>>>> I’ve tried deleting the db files but no difference >>>>> unfortunately. Starting manually or even just restarting >>>>> the service works, so I believe the XML is all as it >>>>> should be. This is my output from systemctl when it’s in >>>>> this ‘hung' state: >>>>> >>>>> ●freeswitch.service - freeswitch >>>>> Loaded: loaded (/etc/systemd/system/freeswitch.service; >>>>> enabled) >>>>> Active: active (running)since Wed 2017-06-28 14:36:23 >>>>> BST; 3min 27s ago >>>>> Process: 452 ExecStart=/usr/bin/freeswitch -ncwait >>>>> -nonat (code=exited, status=0/SUCCESS) >>>>> Main PID: 472 (freeswitch) >>>>> CGroup: /system.slice/freeswitch.service >>>>> └─472 /usr/bin/freeswitch -ncwait -nonat >>>>> >>>>> Jun 28 14:36:20 server systemd[1]: Starting freeswitch... >>>>> Jun 28 14:36:20 server freeswitch[452]: 472 Backgrounding. >>>>> Jun 28 14:36:23 server freeswitch[452]: FreeSWITCH[452] >>>>> Waiting for background process pid:472 to be ready..... >>>>> Jun 28 14:36:23 server freeswitch[452]: FreeSWITCH[452] >>>>> System Ready pid:472 >>>>> Jun 28 14:36:23 server systemd[1]: Started freeswitch. >>>>> >>>>> >>>>>> On 28 Jun 2017, at 14:18, Mundkowsky, Robert >>>>>> > wrote: >>>>>> >>>>>> Yes. I think we had a similar problem in ,I think, two >>>>>> situations. >>>>>> 1)The file system database can get messed up. >>>>>> a.Stop FS >>>>>> b.You can clear then via rm >>>>>> /export/Apps/freeswitch/var/lib/freeswitch/db/* >>>>>> c.Restart FS >>>>>> 2)If there is an error in the configuration files. >>>>>> a.Take a look at >>>>>> /export/Apps/freeswitch/var/log/freeswitch/freeswitch.xml.fsxml >>>>>> to see if there are any errors in XML. >>>>>> You can get more details of start problem via: >>>>>> journalctl -xn >>>>>> systemctl status freeswitch.service >>>>>> if you do not get enough details, try starting FS >>>>>> directly and you might get more details via: >>>>>> /…/bin/freeswitch >>>>>> Robert Mundkowsky >>>>>> *From:*FreeSWITCH-users >>>>>> [mailto:freeswitch-users-bounces at lists.freeswitch.org >>>>>> ]*On >>>>>> Behalf Of*Rick Jarvis >>>>>> *Sent:*Wednesday, June 28, 2017 7:44 AM >>>>>> *To:*FreeSWITCH Users Help >>>>>> >>>>> > >>>>>> *Subject:*[Freeswitch-users] FS service issues (systemd) >>>>>> >>>>>> Got a weird issue where FS isn’t starting up correctly. >>>>>> Looking at the logs, it appears just to be attempting to >>>>>> restart SIP profiles (and failing), and that’s about it. >>>>>> There’s no socket for fs_cli, and a simple ‘systemctl >>>>>> restart’ fixes it. Anyone had something like this? I’ve >>>>>> tried fixing it with systemd after / requires, but it >>>>>> hasn’t helped. >>>>>> Here’s my .service file (not sure if this is the issue, >>>>>> I'd welcome any comments / better versions?) >>>>>> [Unit] >>>>>> Description=freeswitch >>>>>> After=syslog.target network-online.target local-fs.target >>>>>> [Service] >>>>>> ; service >>>>>> Type=forking >>>>>> PIDFile=/run/freeswitch/freeswitch.pid >>>>>> PermissionsStartOnly=true >>>>>> ExecStart=/usr/bin/freeswitch -ncwait -nonat >>>>>> ExecStop=/usr/bin/fs_cli -x shutdown >>>>>> TimeoutSec=45s >>>>>> Restart=always >>>>>> ; exec >>>>>> RuntimeDirectory=freeswitch >>>>>> RuntimeDirectoryMode=0755 >>>>>> User=freeswitch >>>>>> Group=freeswitch >>>>>> LimitCORE=infinity >>>>>> LimitNOFILE=100000 >>>>>> LimitNPROC=60000 >>>>>> ;LimitSTACK=240 >>>>>> LimitRTPRIO=infinity >>>>>> LimitRTTIME=7000000 >>>>>> IOSchedulingClass=realtime >>>>>> IOSchedulingPriority=2 >>>>>> CPUSchedulingPolicy=rr >>>>>> CPUSchedulingPriority=89 >>>>>> UMask=0007 >>>>>> [Install] >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> ------------------------------------------------------------------------ >>>>>> >>>>>> This e-mail and any files transmitted with it may contain >>>>>> privileged or confidential information. It is solely for >>>>>> use by the individual for whom it is intended, even if >>>>>> addressed incorrectly. If you received this e-mail in >>>>>> error, please notify the sender; do not disclose, copy, >>>>>> distribute, or take any action in reliance on the >>>>>> contents of this information; and delete it from your >>>>>> system. Any other use of this e-mail is prohibited. >>>>>> >>>>>> >>>>>> Thank you for your compliance. >>>>>> >>>>>> ------------------------------------------------------------------------ >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 >>> OpenTelecom.IT >>> cell: +39 347 266 56 18 >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at 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 at 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 at 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 at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From rick at magicmail.mooo.com Wed Jun 28 20:00:32 2017 From: rick at magicmail.mooo.com (Rick Jarvis) Date: Wed, 28 Jun 2017 21:00:32 +0100 Subject: [Freeswitch-users] FS service issues (systemd) In-Reply-To: <0a5201d2f047$fa8f8660$efae9320$@freeswitch.org> References: <6B172391-E75C-4CA2-A2DE-2C0A652C19EC@magicmail.mooo.com> <518D9916-4387-4C50-9932-DE812A12856B@magicmail.mooo.com> <4773C447-BBDD-46BC-8584-4A776A726FF5@gmail.com> <828993D4-5964-4EE7-87B6-D15830E045D3@magicmail.mooo.com> <0a5201d2f047$fa8f8660$efae9320$@freeswitch.org> Message-ID: <1EC5CAF7-6C05-4F94-A057-4FA94E961C58@magicmail.mooo.com> Got that, it’s this in the service file: [Unit] Description=freeswitch After=network-online.target > On 28 Jun 2017, at 20:51, Ken Rice wrote: > > Try changing it so that FreeSWITCH depends on the network to be full up before it’ll start in the system setup. Theres a way to do that but the data eludes me at this time > > From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org ] On Behalf Of Rick Jarvis > Sent: Wednesday, June 28, 2017 2:49 PM > To: FreeSWITCH Users Help > > Subject: Re: [Freeswitch-users] FS service issues (systemd) > > Thanks Jason. > > Indeed, fresh install of deb 8.8 and FS 1.6, and exactly the same behaviour. Needs a manual ‘systemctl restart freeswitch’ to get it into life. Quite annoying, especially as it was ok before… not quite sure what’s changed. > >> On 28 Jun 2017, at 20:26, Jason Komar > wrote: >> >> I had this happen on one particular FS box and no matter what I tried I could not get FS to wait until the NIC was 100% up and running. I ended up installing a different model NIC and it started working. I can't remember what model the troublesome one was, unfortunately. >> >> Jason >> >> >> On Wed, Jun 28, 2017 at 8:54 AM, Rick Jarvis > wrote: >>> Debian Jessie, FreeSWITCH Version 1.4.26-37~64bit (-37 64bit) >>> >>> It starts fine manually, and works if I restart it with systemctl restart, which is why I’m confused, but leaning towards it being a dependency not having started in time when it boots. I can probably fix it by reinstalling and/or upgrading, I’d just like to know why it’s failing ideally... >>> >>> >>>> On 28 Jun 2017, at 15:44, Giovanni Maruzzelli > wrote: >>>> >>>> What exact platform are you using? Os/distro? FS version? Any virtualization? >>>> >>>> Has ever started correctly? >>>> >>>> Have you started it from command line (maybe as root user) and then the problems started to appear? >>>> >>>> Have you tried to reinstall from scratch following strictly the confluence instructions? >>>> >>>> Maybe file/directories permission problem? >>>> >>>> -giovanni >>>> >>>> On 28 June 2017 at 16:32, Rick Jarvis > wrote: >>>>> Just looping with failed registrations: >>>>> >>>>> 2017-06-28 15:29:42.738155 [NOTICE] sofia_reg.c:448 Registering connection1 >>>>> 2017-06-28 15:29:42.738155 [NOTICE] sofia_reg.c:448 Registering connection2 >>>>> 2017-06-28 15:29:42.738155 [NOTICE] sofia_reg.c:448 Registering xprov >>>>> 2017-06-28 15:29:42.738155 [ERR] sofia_reg.c:2392 connection2 Failed Registration with status Service Unavailable [503]. failure #101 >>>>> 2017-06-28 15:29:42.738155 [ERR] sofia_reg.c:2392 xprov Failed Registration with status Service Unavailable [503]. failure #101 >>>>> 2017-06-28 15:29:42.838117 [ERR] sofia_reg.c:2392 connection1 Failed Registration with status Service Unavailable [503]. failure #101 >>>>> 2017-06-28 15:29:43.738108 [WARNING] sofia_reg.c:505 connection1 Failed Registration [503], setting retry to 30 seconds. >>>>> 2017-06-28 15:29:43.738108 [WARNING] sofia_reg.c:505 connection2 Failed Registration [503], setting retry to 30 seconds. >>>>> 2017-06-28 15:29:43.738108 [WARNING] sofia_reg.c:505 xprov Failed Registration [503], setting retry to 30 seconds. >>>>> >>>>>> On 28 Jun 2017, at 14:52, Luis Jimenez > wrote: >>>>>> >>>>>> What about the freeswitch.log file when fs is hung? >>>>>> >>>>>> >>>>>> On Jun 28, 2017, at 09:42, Rick Jarvis > wrote: >>>>>> >>>>>>> I’ve tried deleting the db files but no difference unfortunately. Starting manually or even just restarting the service works, so I believe the XML is all as it should be. This is my output from systemctl when it’s in this ‘hung' state: >>>>>>> >>>>>>> ● freeswitch.service - freeswitch >>>>>>> Loaded: loaded (/etc/systemd/system/freeswitch.service; enabled) >>>>>>> Active: active (running) since Wed 2017-06-28 14:36:23 BST; 3min 27s ago >>>>>>> Process: 452 ExecStart=/usr/bin/freeswitch -ncwait -nonat (code=exited, status=0/SUCCESS) >>>>>>> Main PID: 472 (freeswitch) >>>>>>> CGroup: /system.slice/freeswitch.service >>>>>>> └─472 /usr/bin/freeswitch -ncwait -nonat >>>>>>> >>>>>>> Jun 28 14:36:20 server systemd[1]: Starting freeswitch... >>>>>>> Jun 28 14:36:20 server freeswitch[452]: 472 Backgrounding. >>>>>>> Jun 28 14:36:23 server freeswitch[452]: FreeSWITCH[452] Waiting for background process pid:472 to be ready..... >>>>>>> Jun 28 14:36:23 server freeswitch[452]: FreeSWITCH[452] System Ready pid:472 >>>>>>> Jun 28 14:36:23 server systemd[1]: Started freeswitch. >>>>>>> >>>>>>> >>>>>>>> On 28 Jun 2017, at 14:18, Mundkowsky, Robert > wrote: >>>>>>>> >>>>>>>> Yes. I think we had a similar problem in ,I think, two situations. <> >>>>>>>> >>>>>>>> 1) The file system database can get messed up. >>>>>>>> a. Stop FS >>>>>>>> b. You can clear then via rm /export/Apps/freeswitch/var/lib/freeswitch/db/* >>>>>>>> c. Restart FS >>>>>>>> 2) If there is an error in the configuration files. >>>>>>>> a. Take a look at /export/Apps/freeswitch/var/log/freeswitch/freeswitch.xml.fsxml to see if there are any errors in XML. >>>>>>>> >>>>>>>> You can get more details of start problem via: >>>>>>>> journalctl -xn >>>>>>>> systemctl status freeswitch.service >>>>>>>> >>>>>>>> if you do not get enough details, try starting FS directly and you might get more details via: >>>>>>>> >>>>>>>> /…/bin/freeswitch >>>>>>>> >>>>>>>> >>>>>>>> Robert Mundkowsky >>>>>>>> >>>>>>>> From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org ] On Behalf Of Rick Jarvis >>>>>>>> Sent: Wednesday, June 28, 2017 7:44 AM >>>>>>>> To: FreeSWITCH Users Help > >>>>>>>> Subject: [Freeswitch-users] FS service issues (systemd) >>>>>>>> >>>>>>>> >>>>>>>> Got a weird issue where FS isn’t starting up correctly. Looking at the logs, it appears just to be attempting to restart SIP profiles (and failing), and that’s about it. There’s no socket for fs_cli, and a simple ‘systemctl restart’ fixes it. Anyone had something like this? I’ve tried fixing it with systemd after / requires, but it hasn’t helped. >>>>>>>> >>>>>>>> Here’s my .service file (not sure if this is the issue, I'd welcome any comments / better versions?) >>>>>>>> >>>>>>>> [Unit] >>>>>>>> Description=freeswitch >>>>>>>> After=syslog.target network-online.target local-fs.target >>>>>>>> >>>>>>>> [Service] >>>>>>>> ; service >>>>>>>> Type=forking >>>>>>>> PIDFile=/run/freeswitch/freeswitch.pid >>>>>>>> PermissionsStartOnly=true >>>>>>>> ExecStart=/usr/bin/freeswitch -ncwait -nonat >>>>>>>> ExecStop=/usr/bin/fs_cli -x shutdown >>>>>>>> TimeoutSec=45s >>>>>>>> Restart=always >>>>>>>> ; exec >>>>>>>> RuntimeDirectory=freeswitch >>>>>>>> RuntimeDirectoryMode=0755 >>>>>>>> User=freeswitch >>>>>>>> Group=freeswitch >>>>>>>> LimitCORE=infinity >>>>>>>> LimitNOFILE=100000 >>>>>>>> LimitNPROC=60000 >>>>>>>> ;LimitSTACK=240 >>>>>>>> LimitRTPRIO=infinity >>>>>>>> LimitRTTIME=7000000 >>>>>>>> IOSchedulingClass=realtime >>>>>>>> IOSchedulingPriority=2 >>>>>>>> CPUSchedulingPolicy=rr >>>>>>>> CPUSchedulingPriority=89 >>>>>>>> UMask=0007 >>>>>>>> >>>>>>>> [Install] >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> >>>>>>>> This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited. >>>>>>>> >>>>>>>> >>>>>>>> Thank you for your compliance. >>>>>>>> _________________________________________________________________________ >>>>>>>> Professional FreeSWITCH Consulting Services: >>>>>>>> consulting at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 >>>> OpenTelecom.IT >>>> cell: +39 347 266 56 18 >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at 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 at 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 at 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 at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From rmundkowsky at ets.org Wed Jun 28 20:01:56 2017 From: rmundkowsky at ets.org (Mundkowsky, Robert) Date: Wed, 28 Jun 2017 20:01:56 +0000 Subject: [Freeswitch-users] FS service issues (systemd) In-Reply-To: <0a5201d2f047$fa8f8660$efae9320$@freeswitch.org> References: <6B172391-E75C-4CA2-A2DE-2C0A652C19EC@magicmail.mooo.com> <518D9916-4387-4C50-9932-DE812A12856B@magicmail.mooo.com> <4773C447-BBDD-46BC-8584-4A776A726FF5@gmail.com> <828993D4-5964-4EE7-87B6-D15830E045D3@magicmail.mooo.com> <0a5201d2f047$fa8f8660$efae9320$@freeswitch.org> Message-ID: That is an interesting idea. Might be like https://stackoverflow.com/questions/21830670/systemd-start-service-after-specific-service After=syslog.target network.target Robert From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ken Rice Sent: Wednesday, June 28, 2017 3:52 PM To: 'FreeSWITCH Users Help' Subject: Re: [Freeswitch-users] FS service issues (systemd) Try changing it so that FreeSWITCH depends on the network to be full up before it’ll start in the system setup. Theres a way to do that but the data eludes me at this time From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rick Jarvis Sent: Wednesday, June 28, 2017 2:49 PM To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] FS service issues (systemd) Thanks Jason. Indeed, fresh install of deb 8.8 and FS 1.6, and exactly the same behaviour. Needs a manual ‘systemctl restart freeswitch’ to get it into life. Quite annoying, especially as it was ok before… not quite sure what’s changed. On 28 Jun 2017, at 20:26, Jason Komar > wrote: I had this happen on one particular FS box and no matter what I tried I could not get FS to wait until the NIC was 100% up and running. I ended up installing a different model NIC and it started working. I can't remember what model the troublesome one was, unfortunately. Jason On Wed, Jun 28, 2017 at 8:54 AM, Rick Jarvis > wrote: Debian Jessie, FreeSWITCH Version 1.4.26-37~64bit (-37 64bit) It starts fine manually, and works if I restart it with systemctl restart, which is why I’m confused, but leaning towards it being a dependency not having started in time when it boots. I can probably fix it by reinstalling and/or upgrading, I’d just like to know why it’s failing ideally... On 28 Jun 2017, at 15:44, Giovanni Maruzzelli > wrote: What exact platform are you using? Os/distro? FS version? Any virtualization? Has ever started correctly? Have you started it from command line (maybe as root user) and then the problems started to appear? Have you tried to reinstall from scratch following strictly the confluence instructions? Maybe file/directories permission problem? -giovanni On 28 June 2017 at 16:32, Rick Jarvis > wrote: Just looping with failed registrations: 2017-06-28 15:29:42.738155 [NOTICE] sofia_reg.c:448 Registering connection1 2017-06-28 15:29:42.738155 [NOTICE] sofia_reg.c:448 Registering connection2 2017-06-28 15:29:42.738155 [NOTICE] sofia_reg.c:448 Registering xprov 2017-06-28 15:29:42.738155 [ERR] sofia_reg.c:2392 connection2 Failed Registration with status Service Unavailable [503]. failure #101 2017-06-28 15:29:42.738155 [ERR] sofia_reg.c:2392 xprov Failed Registration with status Service Unavailable [503]. failure #101 2017-06-28 15:29:42.838117 [ERR] sofia_reg.c:2392 connection1 Failed Registration with status Service Unavailable [503]. failure #101 2017-06-28 15:29:43.738108 [WARNING] sofia_reg.c:505 connection1 Failed Registration [503], setting retry to 30 seconds. 2017-06-28 15:29:43.738108 [WARNING] sofia_reg.c:505 connection2 Failed Registration [503], setting retry to 30 seconds. 2017-06-28 15:29:43.738108 [WARNING] sofia_reg.c:505 xprov Failed Registration [503], setting retry to 30 seconds. On 28 Jun 2017, at 14:52, Luis Jimenez > wrote: What about the freeswitch.log file when fs is hung? On Jun 28, 2017, at 09:42, Rick Jarvis > wrote: I’ve tried deleting the db files but no difference unfortunately. Starting manually or even just restarting the service works, so I believe the XML is all as it should be. This is my output from systemctl when it’s in this ‘hung' state: ● freeswitch.service - freeswitch Loaded: loaded (/etc/systemd/system/freeswitch.service; enabled) Active: active (running) since Wed 2017-06-28 14:36:23 BST; 3min 27s ago Process: 452 ExecStart=/usr/bin/freeswitch -ncwait -nonat (code=exited, status=0/SUCCESS) Main PID: 472 (freeswitch) CGroup: /system.slice/freeswitch.service └─472 /usr/bin/freeswitch -ncwait -nonat Jun 28 14:36:20 server systemd[1]: Starting freeswitch... Jun 28 14:36:20 server freeswitch[452]: 472 Backgrounding. Jun 28 14:36:23 server freeswitch[452]: FreeSWITCH[452] Waiting for background process pid:472 to be ready..... Jun 28 14:36:23 server freeswitch[452]: FreeSWITCH[452] System Ready pid:472 Jun 28 14:36:23 server systemd[1]: Started freeswitch. On 28 Jun 2017, at 14:18, Mundkowsky, Robert > wrote: Yes. I think we had a similar problem in ,I think, two situations. 1) The file system database can get messed up. a. Stop FS b. You can clear then via rm /export/Apps/freeswitch/var/lib/freeswitch/db/* c. Restart FS 2) If there is an error in the configuration files. a. Take a look at /export/Apps/freeswitch/var/log/freeswitch/freeswitch.xml.fsxml to see if there are any errors in XML. You can get more details of start problem via: journalctl -xn systemctl status freeswitch.service if you do not get enough details, try starting FS directly and you might get more details via: /…/bin/freeswitch Robert Mundkowsky From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rick Jarvis Sent: Wednesday, June 28, 2017 7:44 AM To: FreeSWITCH Users Help > Subject: [Freeswitch-users] FS service issues (systemd) Got a weird issue where FS isn’t starting up correctly. Looking at the logs, it appears just to be attempting to restart SIP profiles (and failing), and that’s about it. There’s no socket for fs_cli, and a simple ‘systemctl restart’ fixes it. Anyone had something like this? I’ve tried fixing it with systemd after / requires, but it hasn’t helped. Here’s my .service file (not sure if this is the issue, I'd welcome any comments / better versions?) [Unit] Description=freeswitch After=syslog.target network-online.target local-fs.target [Service] ; service Type=forking PIDFile=/run/freeswitch/freeswitch.pid PermissionsStartOnly=true ExecStart=/usr/bin/freeswitch -ncwait -nonat ExecStop=/usr/bin/fs_cli -x shutdown TimeoutSec=45s Restart=always ; exec RuntimeDirectory=freeswitch RuntimeDirectoryMode=0755 User=freeswitch Group=freeswitch LimitCORE=infinity LimitNOFILE=100000 LimitNPROC=60000 ;LimitSTACK=240 LimitRTPRIO=infinity LimitRTTIME=7000000 IOSchedulingClass=realtime IOSchedulingPriority=2 CPUSchedulingPolicy=rr CPUSchedulingPriority=89 UMask=0007 [Install] ________________________________ This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited. Thank you for your compliance. ________________________________ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 OpenTelecom.IT cell: +39 347 266 56 18 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at 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 at 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 at 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 at 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 at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ________________________________ This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited. Thank you for your compliance. ________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL: From rmundkowsky at ets.org Wed Jun 28 20:09:57 2017 From: rmundkowsky at ets.org (Mundkowsky, Robert) Date: Wed, 28 Jun 2017 20:09:57 +0000 Subject: [Freeswitch-users] FS service issues (systemd) In-Reply-To: <1EC5CAF7-6C05-4F94-A057-4FA94E961C58@magicmail.mooo.com> References: <6B172391-E75C-4CA2-A2DE-2C0A652C19EC@magicmail.mooo.com> <518D9916-4387-4C50-9932-DE812A12856B@magicmail.mooo.com> <4773C447-BBDD-46BC-8584-4A776A726FF5@gmail.com> <828993D4-5964-4EE7-87B6-D15830E045D3@magicmail.mooo.com> <0a5201d2f047$fa8f8660$efae9320$@freeswitch.org> <1EC5CAF7-6C05-4F94-A057-4FA94E961C58@magicmail.mooo.com> Message-ID: Ah, bummer, just noted your original already had that. Robert Mundkowsky From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rick Jarvis Sent: Wednesday, June 28, 2017 4:01 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] FS service issues (systemd) Got that, it’s this in the service file: [Unit] Description=freeswitch After=network-online.target On 28 Jun 2017, at 20:51, Ken Rice > wrote: Try changing it so that FreeSWITCH depends on the network to be full up before it’ll start in the system setup. Theres a way to do that but the data eludes me at this time From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rick Jarvis Sent: Wednesday, June 28, 2017 2:49 PM To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] FS service issues (systemd) Thanks Jason. Indeed, fresh install of deb 8.8 and FS 1.6, and exactly the same behaviour. Needs a manual ‘systemctl restart freeswitch’ to get it into life. Quite annoying, especially as it was ok before… not quite sure what’s changed. On 28 Jun 2017, at 20:26, Jason Komar > wrote: I had this happen on one particular FS box and no matter what I tried I could not get FS to wait until the NIC was 100% up and running. I ended up installing a different model NIC and it started working. I can't remember what model the troublesome one was, unfortunately. Jason On Wed, Jun 28, 2017 at 8:54 AM, Rick Jarvis > wrote: Debian Jessie, FreeSWITCH Version 1.4.26-37~64bit (-37 64bit) It starts fine manually, and works if I restart it with systemctl restart, which is why I’m confused, but leaning towards it being a dependency not having started in time when it boots. I can probably fix it by reinstalling and/or upgrading, I’d just like to know why it’s failing ideally... On 28 Jun 2017, at 15:44, Giovanni Maruzzelli > wrote: What exact platform are you using? Os/distro? FS version? Any virtualization? Has ever started correctly? Have you started it from command line (maybe as root user) and then the problems started to appear? Have you tried to reinstall from scratch following strictly the confluence instructions? Maybe file/directories permission problem? -giovanni On 28 June 2017 at 16:32, Rick Jarvis > wrote: Just looping with failed registrations: 2017-06-28 15:29:42.738155 [NOTICE] sofia_reg.c:448 Registering connection1 2017-06-28 15:29:42.738155 [NOTICE] sofia_reg.c:448 Registering connection2 2017-06-28 15:29:42.738155 [NOTICE] sofia_reg.c:448 Registering xprov 2017-06-28 15:29:42.738155 [ERR] sofia_reg.c:2392 connection2 Failed Registration with status Service Unavailable [503]. failure #101 2017-06-28 15:29:42.738155 [ERR] sofia_reg.c:2392 xprov Failed Registration with status Service Unavailable [503]. failure #101 2017-06-28 15:29:42.838117 [ERR] sofia_reg.c:2392 connection1 Failed Registration with status Service Unavailable [503]. failure #101 2017-06-28 15:29:43.738108 [WARNING] sofia_reg.c:505 connection1 Failed Registration [503], setting retry to 30 seconds. 2017-06-28 15:29:43.738108 [WARNING] sofia_reg.c:505 connection2 Failed Registration [503], setting retry to 30 seconds. 2017-06-28 15:29:43.738108 [WARNING] sofia_reg.c:505 xprov Failed Registration [503], setting retry to 30 seconds. On 28 Jun 2017, at 14:52, Luis Jimenez > wrote: What about the freeswitch.log file when fs is hung? On Jun 28, 2017, at 09:42, Rick Jarvis > wrote: I’ve tried deleting the db files but no difference unfortunately. Starting manually or even just restarting the service works, so I believe the XML is all as it should be. This is my output from systemctl when it’s in this ‘hung' state: ● freeswitch.service - freeswitch Loaded: loaded (/etc/systemd/system/freeswitch.service; enabled) Active: active (running) since Wed 2017-06-28 14:36:23 BST; 3min 27s ago Process: 452 ExecStart=/usr/bin/freeswitch -ncwait -nonat (code=exited, status=0/SUCCESS) Main PID: 472 (freeswitch) CGroup: /system.slice/freeswitch.service └─472 /usr/bin/freeswitch -ncwait -nonat Jun 28 14:36:20 server systemd[1]: Starting freeswitch... Jun 28 14:36:20 server freeswitch[452]: 472 Backgrounding. Jun 28 14:36:23 server freeswitch[452]: FreeSWITCH[452] Waiting for background process pid:472 to be ready..... Jun 28 14:36:23 server freeswitch[452]: FreeSWITCH[452] System Ready pid:472 Jun 28 14:36:23 server systemd[1]: Started freeswitch. On 28 Jun 2017, at 14:18, Mundkowsky, Robert > wrote: Yes. I think we had a similar problem in ,I think, two situations. 1) The file system database can get messed up. a. Stop FS b. You can clear then via rm /export/Apps/freeswitch/var/lib/freeswitch/db/* c. Restart FS 2) If there is an error in the configuration files. a. Take a look at /export/Apps/freeswitch/var/log/freeswitch/freeswitch.xml.fsxml to see if there are any errors in XML. You can get more details of start problem via: journalctl -xn systemctl status freeswitch.service if you do not get enough details, try starting FS directly and you might get more details via: /…/bin/freeswitch Robert Mundkowsky From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rick Jarvis Sent: Wednesday, June 28, 2017 7:44 AM To: FreeSWITCH Users Help > Subject: [Freeswitch-users] FS service issues (systemd) Got a weird issue where FS isn’t starting up correctly. Looking at the logs, it appears just to be attempting to restart SIP profiles (and failing), and that’s about it. There’s no socket for fs_cli, and a simple ‘systemctl restart’ fixes it. Anyone had something like this? I’ve tried fixing it with systemd after / requires, but it hasn’t helped. Here’s my .service file (not sure if this is the issue, I'd welcome any comments / better versions?) [Unit] Description=freeswitch After=syslog.target network-online.target local-fs.target [Service] ; service Type=forking PIDFile=/run/freeswitch/freeswitch.pid PermissionsStartOnly=true ExecStart=/usr/bin/freeswitch -ncwait -nonat ExecStop=/usr/bin/fs_cli -x shutdown TimeoutSec=45s Restart=always ; exec RuntimeDirectory=freeswitch RuntimeDirectoryMode=0755 User=freeswitch Group=freeswitch LimitCORE=infinity LimitNOFILE=100000 LimitNPROC=60000 ;LimitSTACK=240 LimitRTPRIO=infinity LimitRTTIME=7000000 IOSchedulingClass=realtime IOSchedulingPriority=2 CPUSchedulingPolicy=rr CPUSchedulingPriority=89 UMask=0007 [Install] ________________________________ This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited. Thank you for your compliance. ________________________________ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 OpenTelecom.IT cell: +39 347 266 56 18 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at 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 at 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 at 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 at 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 at 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 at 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 at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ________________________________ This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited. Thank you for your compliance. ________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL: From anthony.minessale at gmail.com Wed Jun 28 20:39:18 2017 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 28 Jun 2017 15:39:18 -0500 Subject: [Freeswitch-users] FreeSWITCH and OpenSIPS Training will be available this August at the end of ClueCon Message-ID: There is limited availability so sign up early! https://www.cluecon.com/training.html -- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ * ClueCon Weekly Development Call ☎ sip:888 at conference.freeswitch.org ☎ +19193869900 https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA -------------- next part -------------- An HTML attachment was scrubbed... URL: From rick at magicmail.mooo.com Wed Jun 28 23:15:13 2017 From: rick at magicmail.mooo.com (Rick Jarvis) Date: Thu, 29 Jun 2017 00:15:13 +0100 Subject: [Freeswitch-users] FS service issues (systemd) In-Reply-To: References: <6B172391-E75C-4CA2-A2DE-2C0A652C19EC@magicmail.mooo.com> <518D9916-4387-4C50-9932-DE812A12856B@magicmail.mooo.com> <4773C447-BBDD-46BC-8584-4A776A726FF5@gmail.com> <828993D4-5964-4EE7-87B6-D15830E045D3@magicmail.mooo.com> <0a5201d2f047$fa8f8660$efae9320$@freeswitch.org> <1EC5CAF7-6C05-4F94-A057-4FA94E961C58@magicmail.mooo.com> Message-ID: <48C48764-8B13-4410-94B3-E0EBC2A936A3@magicmail.mooo.com> Cool. Love the way the lack of a background colour makes the bit I highlighted almost invisible ;) FTR: After=network-online.target > On 28 Jun 2017, at 21:09, Mundkowsky, Robert wrote: > > Ah, bummer, just noted your original already had that. > > Robert Mundkowsky > > From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rick Jarvis > Sent: Wednesday, June 28, 2017 4:01 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] FS service issues (systemd) > > Got that, it’s this in the service file: > > [Unit] > Description=freeswitch > After=network-online.target > > On 28 Jun 2017, at 20:51, Ken Rice wrote: > > Try changing it so that FreeSWITCH depends on the network to be full up before it’ll start in the system setup. Theres a way to do that but the data eludes me at this time > > From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rick Jarvis > Sent: Wednesday, June 28, 2017 2:49 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] FS service issues (systemd) > > Thanks Jason. > > Indeed, fresh install of deb 8.8 and FS 1.6, and exactly the same behaviour. Needs a manual ‘systemctl restart freeswitch’ to get it into life. Quite annoying, especially as it was ok before… not quite sure what’s changed. > > On 28 Jun 2017, at 20:26, Jason Komar wrote: > > I had this happen on one particular FS box and no matter what I tried I could not get FS to wait until the NIC was 100% up and running. I ended up installing a different model NIC and it started working. I can't remember what model the troublesome one was, unfortunately. > > Jason > > > On Wed, Jun 28, 2017 at 8:54 AM, Rick Jarvis wrote: > Debian Jessie, FreeSWITCH Version 1.4.26-37~64bit (-37 64bit) > > It starts fine manually, and works if I restart it with systemctl restart, which is why I’m confused, but leaning towards it being a dependency not having started in time when it boots. I can probably fix it by reinstalling and/or upgrading, I’d just like to know why it’s failing ideally... > > > On 28 Jun 2017, at 15:44, Giovanni Maruzzelli wrote: > > What exact platform are you using? Os/distro? FS version? Any virtualization? > > Has ever started correctly? > > Have you started it from command line (maybe as root user) and then the problems started to appear? > > Have you tried to reinstall from scratch following strictly the confluence instructions? > > Maybe file/directories permission problem? > > -giovanni > > On 28 June 2017 at 16:32, Rick Jarvis wrote: > Just looping with failed registrations: > > 2017-06-28 15:29:42.738155 [NOTICE] sofia_reg.c:448 Registering connection1 > 2017-06-28 15:29:42.738155 [NOTICE] sofia_reg.c:448 Registering connection2 > 2017-06-28 15:29:42.738155 [NOTICE] sofia_reg.c:448 Registering xprov > 2017-06-28 15:29:42.738155 [ERR] sofia_reg.c:2392 connection2 Failed Registration with status Service Unavailable [503]. failure #101 > 2017-06-28 15:29:42.738155 [ERR] sofia_reg.c:2392 xprov Failed Registration with status Service Unavailable [503]. failure #101 > 2017-06-28 15:29:42.838117 [ERR] sofia_reg.c:2392 connection1 Failed Registration with status Service Unavailable [503]. failure #101 > 2017-06-28 15:29:43.738108 [WARNING] sofia_reg.c:505 connection1 Failed Registration [503], setting retry to 30 seconds. > 2017-06-28 15:29:43.738108 [WARNING] sofia_reg.c:505 connection2 Failed Registration [503], setting retry to 30 seconds. > 2017-06-28 15:29:43.738108 [WARNING] sofia_reg.c:505 xprov Failed Registration [503], setting retry to 30 seconds. > > On 28 Jun 2017, at 14:52, Luis Jimenez wrote: > > What about the freeswitch.log file when fs is hung? > > > On Jun 28, 2017, at 09:42, Rick Jarvis wrote: > > I’ve tried deleting the db files but no difference unfortunately. Starting manually or even just restarting the service works, so I believe the XML is all as it should be. This is my output from systemctl when it’s in this ‘hung' state: > > ● freeswitch.service - freeswitch > Loaded: loaded (/etc/systemd/system/freeswitch.service; enabled) > Active: active (running) since Wed 2017-06-28 14:36:23 BST; 3min 27s ago > Process: 452 ExecStart=/usr/bin/freeswitch -ncwait -nonat (code=exited, status=0/SUCCESS) > Main PID: 472 (freeswitch) > CGroup: /system.slice/freeswitch.service > └─472 /usr/bin/freeswitch -ncwait -nonat > > Jun 28 14:36:20 server systemd[1]: Starting freeswitch... > Jun 28 14:36:20 server freeswitch[452]: 472 Backgrounding. > Jun 28 14:36:23 server freeswitch[452]: FreeSWITCH[452] Waiting for background process pid:472 to be ready..... > Jun 28 14:36:23 server freeswitch[452]: FreeSWITCH[452] System Ready pid:472 > Jun 28 14:36:23 server systemd[1]: Started freeswitch. > > > On 28 Jun 2017, at 14:18, Mundkowsky, Robert wrote: > > Yes. I think we had a similar problem in ,I think, two situations. > > 1) The file system database can get messed up. > a. Stop FS > b. You can clear then via rm /export/Apps/freeswitch/var/lib/freeswitch/db/* > c. Restart FS > 2) If there is an error in the configuration files. > a. Take a look at /export/Apps/freeswitch/var/log/freeswitch/freeswitch.xml.fsxml to see if there are any errors in XML. > > You can get more details of start problem via: > journalctl -xn > systemctl status freeswitch.service > > if you do not get enough details, try starting FS directly and you might get more details via: > > /…/bin/freeswitch > > > Robert Mundkowsky > > From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rick Jarvis > Sent: Wednesday, June 28, 2017 7:44 AM > To: FreeSWITCH Users Help > Subject: [Freeswitch-users] FS service issues (systemd) > > > Got a weird issue where FS isn’t starting up correctly. Looking at the logs, it appears just to be attempting to restart SIP profiles (and failing), and that’s about it. There’s no socket for fs_cli, and a simple ‘systemctl restart’ fixes it. Anyone had something like this? I’ve tried fixing it with systemd after / requires, but it hasn’t helped. > > Here’s my .service file (not sure if this is the issue, I'd welcome any comments / better versions?) > > [Unit] > Description=freeswitch > After=syslog.target network-online.target local-fs.target > > [Service] > ; service > Type=forking > PIDFile=/run/freeswitch/freeswitch.pid > PermissionsStartOnly=true > ExecStart=/usr/bin/freeswitch -ncwait -nonat > ExecStop=/usr/bin/fs_cli -x shutdown > TimeoutSec=45s > Restart=always > ; exec > RuntimeDirectory=freeswitch > RuntimeDirectoryMode=0755 > User=freeswitch > Group=freeswitch > LimitCORE=infinity > LimitNOFILE=100000 > LimitNPROC=60000 > ;LimitSTACK=240 > LimitRTPRIO=infinity > LimitRTTIME=7000000 > IOSchedulingClass=realtime > IOSchedulingPriority=2 > CPUSchedulingPolicy=rr > CPUSchedulingPriority=89 > UMask=0007 > > [Install] > > > > > > > > This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited. > > > > Thank you for your compliance. > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 > OpenTelecom.IT > cell: +39 347 266 56 18 > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at 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 at 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 at 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 at 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 at 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 at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited. > > > Thank you for your compliance. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From kkothari157 at gmail.com Thu Jun 29 08:22:41 2017 From: kkothari157 at gmail.com (Ketan Kothari) Date: Thu, 29 Jun 2017 13:52:41 +0530 Subject: [Freeswitch-users] choppy sound of voicemail recording files Message-ID: Hi Guys we have implemented voicemail features on our system but we are facing an issue with choppy sound of recording files. When the files are playing an instructions the sounds is choppy and unable to listen it properly we use "rtp-autofix-timing" variable to resolve this issue and really it works fine for us but then its not allow us to record a message and hang up the call with message "Message is less than minimum record length". -------------- next part -------------- An HTML attachment was scrubbed... URL: From shaun.stokes at itec-support.co.uk Thu Jun 29 09:34:30 2017 From: shaun.stokes at itec-support.co.uk (Shaun Stokes) Date: Thu, 29 Jun 2017 09:34:30 +0000 Subject: [Freeswitch-users] SKYPE: mod_skypopen END-OF-LIFE In-Reply-To: References: Message-ID: <6FD2F8B5BB72834E9939AEDF9FB802A901E8678625@mbx-01.sysconfig.co.uk> Microsoft are shifting their focus to the Cloud and in doing so have stripped down the Skype API so won’t be possible: https://support.skype.com/en/faq/FA214/what-is-the-desktop-api This is a great opportunity for open source solutions such as FreeSWITCH, Jitsi and more to step up to the mark. From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Vladimir Gaitner Sent: 28 June 2017 17:49 To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] SKYPE: mod_skypopen END-OF-LIFE Hello Geowanni, Do you have any planns to rewrite the module? ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ [http://www.itec-support.co.uk/wp-content/uploads/2016/07/email_logo.jpg] Shaun Stokes - Infrastructure Analyst T : 01453 700713 E : shaun.stokes at itec-support.co.uk W : www.itec-support.co.uk Registered Address :- ITEC Support, Suite 2 Prospect House, Bath Road, Stroud, Gloucestershire GL5 3QF Company No. 06908001 CONFIDENTIALITY NOTICE This communication and the information it contains are intended for the person or organisation to which it is addressed. Its contents are confidential and may be protected in law. Unauthorised use, copying or disclosure of any of it may be unlawful. If you are not the intended recipient, please contact us immediately. The contents of any attachments in this e-mail may contain software viruses, which could damage your own computer system. While ITEC Support has taken every reasonable precaution to minimise this risk, we cannot accept liability for any damage which you sustain as a result of software viruses. You should carry out your own virus checking procedure before opening any attachment. ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL: From gmaruzz at gmail.com Thu Jun 29 10:12:03 2017 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Thu, 29 Jun 2017 12:12:03 +0200 Subject: [Freeswitch-users] SKYPE: mod_skypopen END-OF-LIFE In-Reply-To: <6FD2F8B5BB72834E9939AEDF9FB802A901E8678625@mbx-01.sysconfig.co.uk> References: <6FD2F8B5BB72834E9939AEDF9FB802A901E8678625@mbx-01.sysconfig.co.uk> Message-ID: On 29 June 2017 at 11:34, Shaun Stokes wrote: > Microsoft are shifting their focus to the Cloud and in doing so have > stripped down the Skype API so won’t be possible: > > https://support.skype.com/en/faq/FA214/what-is-the-desktop-api > > > > This is a great opportunity for open source solutions such as FreeSWITCH, > Jitsi and more to step up to the mark. > YES, we'll have a lot of nice new things at ClueCon !!! ;) -giovanni > > > *From:* FreeSWITCH-users [mailto:freeswitch-users- > bounces at lists.freeswitch.org] *On Behalf Of *Vladimir Gaitner > *Sent:* 28 June 2017 17:49 > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* [Freeswitch-users] SKYPE: mod_skypopen END-OF-LIFE > > > > Hello Geowanni, > > Do you have any planns to rewrite the module? > > > > > > > > > ______________________________________________________________________ > This message has been checked for all known viruses by MessageLabs Virus > Scanning Service. > ______________________________________________________________________ > Shaun Stokes - Infrastructure Analyst > T : 01453 700713 > E : shaun.stokes at itec-support.co.uk > W : www.itec-support.co.uk > Registered Address :- ITEC Support, Suite 2 Prospect House, Bath Road, > Stroud, Gloucestershire GL5 3QF > Company No. 06908001 > > CONFIDENTIALITY NOTICE > This communication and the information it contains are intended for the > person or organisation to which it is addressed. Its contents are > confidential and may be protected in law. Unauthorised use, copying or > disclosure of any of it may be unlawful. If you are not the intended > recipient, please contact us immediately. > The contents of any attachments in this e-mail may contain software > viruses, which could damage your own computer system. While ITEC Support > has taken every reasonable precaution to minimise this risk, we cannot > accept liability for any damage which you sustain as a result of software > viruses. You should carry out your own virus checking procedure before > opening any attachment. > > ______________________________________________________________________ > This message has been checked for all known viruses by MessageLabs Virus > Scanning Service. > ______________________________________________________________________ > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at 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 OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: From khamlichi.khalil at gmail.com Thu Jun 29 10:48:16 2017 From: khamlichi.khalil at gmail.com (Khalil Khamlichi) Date: Thu, 29 Jun 2017 10:48:16 +0000 Subject: [Freeswitch-users] Callcenter module, can I originate call for an agent in uuid-standby mode ? In-Reply-To: References: <4770F7F5-BCBB-47B6-A909-263EEFEADFE4@jerris.com> <1497890891.572062.1014326656.673B64EF@webmail.messagingengine.com> <8C4372DB-F2EB-4614-82EC-185B5AA33C65@jerris.com> Message-ID: Thanks a lot Italo, this solved my problem. On Wed, Jun 21, 2017 at 12:44 PM, Ítalo Rossi wrote: > uuid-standby is the same as a Consumer agent for mod_fifo (off-hook agent). > > Khalil, > > Yes it's possible, it's just a matter of putting the right commands on > your dialplan. When you log in your uuid-standby agent, put the session > uuid in a hash (hash insert/agents_uuid/myagent/uuid), then you query > this hash (hash select/agents_uuid/myagent) before originate and set a > variable on the originate channel, when the call is established you call > intercept app with the agent uuid: > > > > This will bridge your originated call with the agent's standby leg. You > can even pause the agent before calling intercept to make sure the agent > doesn't get another call or call callcenter_track app > callcenter_track(myagent), this will tell mod_callcenter that this agent > has an external call and the agent won't be called until the number of > external_calls is 0 again. > > I'm pretty sure you can do the same with mod_fifo. > > > > On Tue, Jun 20, 2017 at 12:43 PM, Michael Jerris wrote: > >> Not sure what that means exactly. >> >> On Jun 20, 2017, at 10:00 AM, Khalil Khamlichi < >> khamlichi.khalil at gmail.com> wrote: >> >> alright, does mod_fifo support some sort of uuid_standby mode ? >> >> On Mon, Jun 19, 2017 at 6:27 PM, Michael Jerris wrote: >> >>> mod_fifo has ALWAYS been superior, people assume otherwise because of >>> the name. Check it out, its pretty powerful. mod_callcenter was written >>> because people had a hard time understanding mod_fifo. It supports agent >>> tracking, some skills routing, inbound and outbound agents, etc. If there >>> is stuff missing we should really sort getting it into mod_fifo and abandon >>> mod_callcenter. >>> >>> >>> > On Jun 19, 2017, at 12:48 PM, Michael Avers >>> wrote: >>> > >>> > What makes mod_fifo better these days? Some reason I was under the >>> impression mod_callcenter is a better choice. Can you please give some real >>> world examples where mod_fifo excels compared to mod_callcenter, or >>> features that are not possible to implement with the latter? >>> > >>> > Thanks >>> > Mike >>> > >>> > On Mon, Jun 19, 2017, at 09:34 AM, Michael Jerris wrote: >>> >> mod_fifo is a much more feature rich version of a call queue than >>> mod_callcenter. You might want to check that out instead. >>> >> >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Ítalo Rossi > italo at freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From asilva at wirelessmundi.com Thu Jun 29 11:22:31 2017 From: asilva at wirelessmundi.com (Antonio Silva) Date: Thu, 29 Jun 2017 13:22:31 +0200 Subject: [Freeswitch-users] Error sending queries to pgsql Message-ID: Hi all, i use pgsql in core and from time to time i see critical messages like fail to send query, example: [CRIT] switch_pgsql.c:255 Failed to send query (update sip_authentication set expires='1498726568',last_nc=364 where nonce='3c20da2e-b663-46fe-8edc-c756f9166d07') to database: server closed the connection unexpectedly This was recently, i did an update to current master the previous version was from April, not sure if it could be an error on FS o some other issue on my box.. PG is installed on the same server and the only thing i see from pg is "postgres[2236]: FATAL: invalid frontend message type 21", PG is installed on the same server, running on /dev/shm with the same prio as FS and the process never stopped. anyone has experience this error before? any idea what it could be the cause? Thanks, -- Saludos / Regards / Cumprimentos, António silva From alexandr.popov at iqoption.com Thu Jun 29 12:42:11 2017 From: alexandr.popov at iqoption.com (Alexandr Popov) Date: Thu, 29 Jun 2017 15:42:11 +0300 Subject: [Freeswitch-users] Error sending queries to pgsql In-Reply-To: References: Message-ID: Yeah i get this error 1-2 times everyday. at different instances. 2017-06-29 14:22 GMT+03:00 Antonio Silva : > Hi all, > > i use pgsql in core and from time to time i see critical messages like > fail to send query, example: > > [CRIT] switch_pgsql.c:255 Failed to send query (update sip_authentication > set expires='1498726568',last_nc=364 where nonce='3c20da2e-b663-46fe-8edc-c756f9166d07') > to database: server closed the connection unexpectedly > > This was recently, i did an update to current master the previous version > was from April, not sure if it could be an error on FS o some other issue > on my box.. > > > PG is installed on the same server and the only thing i see from pg is > "postgres[2236]: FATAL: invalid frontend message type 21", PG is installed > on the same server, running on /dev/shm with the same prio as FS and the > process never stopped. > > > anyone has experience this error before? any idea what it could be the > cause? > > > Thanks, > > -- > > Saludos / Regards / Cumprimentos, > António silva > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: From rmundkowsky at ets.org Thu Jun 29 12:40:00 2017 From: rmundkowsky at ets.org (Mundkowsky, Robert) Date: Thu, 29 Jun 2017 12:40:00 +0000 Subject: [Freeswitch-users] FS service issues (systemd) In-Reply-To: <1EC5CAF7-6C05-4F94-A057-4FA94E961C58@magicmail.mooo.com> References: <6B172391-E75C-4CA2-A2DE-2C0A652C19EC@magicmail.mooo.com> <518D9916-4387-4C50-9932-DE812A12856B@magicmail.mooo.com> <4773C447-BBDD-46BC-8584-4A776A726FF5@gmail.com> <828993D4-5964-4EE7-87B6-D15830E045D3@magicmail.mooo.com> <0a5201d2f047$fa8f8660$efae9320$@freeswitch.org> <1EC5CAF7-6C05-4F94-A057-4FA94E961C58@magicmail.mooo.com> Message-ID: So to be clear, you are having a problem starting FS from boot scripts, not from starting it directly from console? We are using I believe SysVinit style boot script for FS rather than systemd. Granted with all the different boot scripts types and overlaps, I might be confused. Robert From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rick Jarvis Sent: Wednesday, June 28, 2017 4:01 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] FS service issues (systemd) Got that, it’s this in the service file: [Unit] Description=freeswitch After=network-online.target On 28 Jun 2017, at 20:51, Ken Rice > wrote: Try changing it so that FreeSWITCH depends on the network to be full up before it’ll start in the system setup. Theres a way to do that but the data eludes me at this time From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rick Jarvis Sent: Wednesday, June 28, 2017 2:49 PM To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] FS service issues (systemd) Thanks Jason. Indeed, fresh install of deb 8.8 and FS 1.6, and exactly the same behaviour. Needs a manual ‘systemctl restart freeswitch’ to get it into life. Quite annoying, especially as it was ok before… not quite sure what’s changed. On 28 Jun 2017, at 20:26, Jason Komar > wrote: I had this happen on one particular FS box and no matter what I tried I could not get FS to wait until the NIC was 100% up and running. I ended up installing a different model NIC and it started working. I can't remember what model the troublesome one was, unfortunately. Jason On Wed, Jun 28, 2017 at 8:54 AM, Rick Jarvis > wrote: Debian Jessie, FreeSWITCH Version 1.4.26-37~64bit (-37 64bit) It starts fine manually, and works if I restart it with systemctl restart, which is why I’m confused, but leaning towards it being a dependency not having started in time when it boots. I can probably fix it by reinstalling and/or upgrading, I’d just like to know why it’s failing ideally... On 28 Jun 2017, at 15:44, Giovanni Maruzzelli > wrote: What exact platform are you using? Os/distro? FS version? Any virtualization? Has ever started correctly? Have you started it from command line (maybe as root user) and then the problems started to appear? Have you tried to reinstall from scratch following strictly the confluence instructions? Maybe file/directories permission problem? -giovanni On 28 June 2017 at 16:32, Rick Jarvis > wrote: Just looping with failed registrations: 2017-06-28 15:29:42.738155 [NOTICE] sofia_reg.c:448 Registering connection1 2017-06-28 15:29:42.738155 [NOTICE] sofia_reg.c:448 Registering connection2 2017-06-28 15:29:42.738155 [NOTICE] sofia_reg.c:448 Registering xprov 2017-06-28 15:29:42.738155 [ERR] sofia_reg.c:2392 connection2 Failed Registration with status Service Unavailable [503]. failure #101 2017-06-28 15:29:42.738155 [ERR] sofia_reg.c:2392 xprov Failed Registration with status Service Unavailable [503]. failure #101 2017-06-28 15:29:42.838117 [ERR] sofia_reg.c:2392 connection1 Failed Registration with status Service Unavailable [503]. failure #101 2017-06-28 15:29:43.738108 [WARNING] sofia_reg.c:505 connection1 Failed Registration [503], setting retry to 30 seconds. 2017-06-28 15:29:43.738108 [WARNING] sofia_reg.c:505 connection2 Failed Registration [503], setting retry to 30 seconds. 2017-06-28 15:29:43.738108 [WARNING] sofia_reg.c:505 xprov Failed Registration [503], setting retry to 30 seconds. On 28 Jun 2017, at 14:52, Luis Jimenez > wrote: What about the freeswitch.log file when fs is hung? On Jun 28, 2017, at 09:42, Rick Jarvis > wrote: I’ve tried deleting the db files but no difference unfortunately. Starting manually or even just restarting the service works, so I believe the XML is all as it should be. This is my output from systemctl when it’s in this ‘hung' state: ● freeswitch.service - freeswitch Loaded: loaded (/etc/systemd/system/freeswitch.service; enabled) Active: active (running) since Wed 2017-06-28 14:36:23 BST; 3min 27s ago Process: 452 ExecStart=/usr/bin/freeswitch -ncwait -nonat (code=exited, status=0/SUCCESS) Main PID: 472 (freeswitch) CGroup: /system.slice/freeswitch.service └─472 /usr/bin/freeswitch -ncwait -nonat Jun 28 14:36:20 server systemd[1]: Starting freeswitch... Jun 28 14:36:20 server freeswitch[452]: 472 Backgrounding. Jun 28 14:36:23 server freeswitch[452]: FreeSWITCH[452] Waiting for background process pid:472 to be ready..... Jun 28 14:36:23 server freeswitch[452]: FreeSWITCH[452] System Ready pid:472 Jun 28 14:36:23 server systemd[1]: Started freeswitch. On 28 Jun 2017, at 14:18, Mundkowsky, Robert > wrote: Yes. I think we had a similar problem in ,I think, two situations. 1) The file system database can get messed up. a. Stop FS b. You can clear then via rm /export/Apps/freeswitch/var/lib/freeswitch/db/* c. Restart FS 2) If there is an error in the configuration files. a. Take a look at /export/Apps/freeswitch/var/log/freeswitch/freeswitch.xml.fsxml to see if there are any errors in XML. You can get more details of start problem via: journalctl -xn systemctl status freeswitch.service if you do not get enough details, try starting FS directly and you might get more details via: /…/bin/freeswitch Robert Mundkowsky From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rick Jarvis Sent: Wednesday, June 28, 2017 7:44 AM To: FreeSWITCH Users Help > Subject: [Freeswitch-users] FS service issues (systemd) Got a weird issue where FS isn’t starting up correctly. Looking at the logs, it appears just to be attempting to restart SIP profiles (and failing), and that’s about it. There’s no socket for fs_cli, and a simple ‘systemctl restart’ fixes it. Anyone had something like this? I’ve tried fixing it with systemd after / requires, but it hasn’t helped. Here’s my .service file (not sure if this is the issue, I'd welcome any comments / better versions?) [Unit] Description=freeswitch After=syslog.target network-online.target local-fs.target [Service] ; service Type=forking PIDFile=/run/freeswitch/freeswitch.pid PermissionsStartOnly=true ExecStart=/usr/bin/freeswitch -ncwait -nonat ExecStop=/usr/bin/fs_cli -x shutdown TimeoutSec=45s Restart=always ; exec RuntimeDirectory=freeswitch RuntimeDirectoryMode=0755 User=freeswitch Group=freeswitch LimitCORE=infinity LimitNOFILE=100000 LimitNPROC=60000 ;LimitSTACK=240 LimitRTPRIO=infinity LimitRTTIME=7000000 IOSchedulingClass=realtime IOSchedulingPriority=2 CPUSchedulingPolicy=rr CPUSchedulingPriority=89 UMask=0007 [Install] ________________________________ This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited. Thank you for your compliance. ________________________________ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at 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 at 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 at 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 at 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 at 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 at 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 at 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 at 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 OpenTelecom.IT cell: +39 347 266 56 18 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at 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 at 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 at 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 at 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 at 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 at 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 at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ________________________________ This e-mail and any files transmitted with it may contain privileged or confidential information. It is solely for use by the individual for whom it is intended, even if addressed incorrectly. If you received this e-mail in error, please notify the sender; do not disclose, copy, distribute, or take any action in reliance on the contents of this information; and delete it from your system. Any other use of this e-mail is prohibited. Thank you for your compliance. ________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL: From daveh at beachdognet.com Thu Jun 29 12:58:42 2017 From: daveh at beachdognet.com (Dave Horton) Date: Thu, 29 Jun 2017 08:58:42 -0400 Subject: [Freeswitch-users] best way to fork media streams for a bridged call? Message-ID: <7BB52F57-983F-4B39-9F58-6E56F4753AF1@beachdognet.com> If I have a bridged call and want to duplicate the two media streams and send each in a separate (one-way, sendonly) SIP dialog to a remote uri, what would be the best way to do that? It looks like it could possibly be achieved using the eavesdrop application (?), or is there an alternate preferred/tested method? From kkothari157 at gmail.com Thu Jun 29 13:10:12 2017 From: kkothari157 at gmail.com (Ketan Kothari) Date: Thu, 29 Jun 2017 18:40:12 +0530 Subject: [Freeswitch-users] Audio issue in Amazon EC2 server Message-ID: Hi All, I have installed freeswitch on Amazon EC2 server and i followed *https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2 * Once we tried to call outbound call getting *no audio in both side* also FreeSWITCH passing my server local IP as RTP-IP to gateway side, =========================== 2017/06/29 12:57:06.579360 128.22.54.32:5060 -> 88.22.45.123:5060 v=0 o=FreeSWITCH 123456456 1245210017 IN IP4 *192.168.1.101* s=FreeSWITCH c=IN IP4 *192.168.1.101* t=0 0 m=audio 31010 RTP/AVP 0 101 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 ============== 128.22.54.32 = Amazon EC2 server public IP *192.168.1.101* = Amazon EC2 server local IP 88.22.45.123 = Gateway provider. Could you please help me on audio issue in local-ip selected as RTP_IP and passing to gateway side? -------------- next part -------------- An HTML attachment was scrubbed... URL: From gmaruzz at gmail.com Thu Jun 29 13:16:38 2017 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Thu, 29 Jun 2017 15:16:38 +0200 Subject: [Freeswitch-users] Audio issue in Amazon EC2 server In-Reply-To: References: Message-ID: On 29 June 2017 at 15:10, Ketan Kothari wrote: > Hi All, > > I have installed freeswitch on Amazon EC2 server and i followed *https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2 > * > > Once we tried to call outbound call getting *no audio in both side* also > FreeSWITCH passing my server local IP as RTP-IP to gateway side, > set your ext-rtp-ip and ext-sip-ip in all your sip_profiles to the EXTERNAL (public) ip address of AWS instance -giovanni -------------- next part -------------- An HTML attachment was scrubbed... URL: From mike at jerris.com Thu Jun 29 16:22:56 2017 From: mike at jerris.com (Michael Jerris) Date: Thu, 29 Jun 2017 12:22:56 -0400 Subject: [Freeswitch-users] Error sending queries to pgsql In-Reply-To: References: Message-ID: <31066AE1-BC31-4C3F-8809-6C5B93C3B0E8@jerris.com> If you can figure out a reliable way to reproduce this issue, please file a jira with details on what causes it. > On Jun 29, 2017, at 7:22 AM, Antonio Silva wrote: > > Hi all, > > i use pgsql in core and from time to time i see critical messages like fail to send query, example: > > [CRIT] switch_pgsql.c:255 Failed to send query (update sip_authentication set expires='1498726568',last_nc=364 where nonce='3c20da2e-b663-46fe-8edc-c756f9166d07') to database: server closed the connection unexpectedly > > This was recently, i did an update to current master the previous version was from April, not sure if it could be an error on FS o some other issue on my box.. > > > PG is installed on the same server and the only thing i see from pg is "postgres[2236]: FATAL: invalid frontend message type 21", PG is installed on the same server, running on /dev/shm with the same prio as FS and the process never stopped. > > > anyone has experience this error before? any idea what it could be the cause? > From mike at jerris.com Thu Jun 29 16:24:00 2017 From: mike at jerris.com (Michael Jerris) Date: Thu, 29 Jun 2017 12:24:00 -0400 Subject: [Freeswitch-users] best way to fork media streams for a bridged call? In-Reply-To: <7BB52F57-983F-4B39-9F58-6E56F4753AF1@beachdognet.com> References: <7BB52F57-983F-4B39-9F58-6E56F4753AF1@beachdognet.com> Message-ID: <6E2456CF-940A-48DB-A28E-E86BC1DC646B@jerris.com> What you describe is basically a spec called SIPREC. We don’t currently support it, but are looking for sponsors. > On Jun 29, 2017, at 8:58 AM, Dave Horton wrote: > > If I have a bridged call and want to duplicate the two media streams and send each in a separate (one-way, sendonly) SIP dialog to a remote uri, what would be the best way to do that? It looks like it could possibly be achieved using the eavesdrop application (?), or is there an alternate preferred/tested method? From asilva at wirelessmundi.com Thu Jun 29 16:31:28 2017 From: asilva at wirelessmundi.com (Antonio Silva) Date: Thu, 29 Jun 2017 18:31:28 +0200 Subject: [Freeswitch-users] Error sending queries to pgsql In-Reply-To: <31066AE1-BC31-4C3F-8809-6C5B93C3B0E8@jerris.com> References: <31066AE1-BC31-4C3F-8809-6C5B93C3B0E8@jerris.com> Message-ID: Hi Michael, Yes, i'm trying to figure it out if is an issue in FS or external.. but the message from PG i can't translate it.. i just enable more logs to see if i got extra hints... Thanks. Saludos / Regards / Cumprimentos, António silva On 06/29/2017 06:22 PM, Michael Jerris wrote: > If you can figure out a reliable way to reproduce this issue, please file a jira with details on what causes it. > >> On Jun 29, 2017, at 7:22 AM, Antonio Silva wrote: >> >> Hi all, >> >> i use pgsql in core and from time to time i see critical messages like fail to send query, example: >> >> [CRIT] switch_pgsql.c:255 Failed to send query (update sip_authentication set expires='1498726568',last_nc=364 where nonce='3c20da2e-b663-46fe-8edc-c756f9166d07') to database: server closed the connection unexpectedly >> >> This was recently, i did an update to current master the previous version was from April, not sure if it could be an error on FS o some other issue on my box.. >> >> >> PG is installed on the same server and the only thing i see from pg is "postgres[2236]: FATAL: invalid frontend message type 21", PG is installed on the same server, running on /dev/shm with the same prio as FS and the process never stopped. >> >> >> anyone has experience this error before? any idea what it could be the cause? >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From s.safarov at gmail.com Fri Jun 30 07:40:10 2017 From: s.safarov at gmail.com (Sergey Safarov) Date: Fri, 30 Jun 2017 07:40:10 +0000 Subject: [Freeswitch-users] may be fix DNS? Message-ID: [safarov at safarov ~]$ ping files.freeswitch.org PING files.freeswitch.org(2607:f348:1021::7 (2607:f348:1021::7)) 56 data bytes >From 2a05:ddc0::48 (2a05:ddc0::48) icmp_seq=1 Destination unreachable: No route >From 2a05:ddc0::48 (2a05:ddc0::48) icmp_seq=2 Destination unreachable: No route -------------- next part -------------- An HTML attachment was scrubbed... URL: From johnnarduchi at gmail.com Thu Jun 29 23:58:37 2017 From: johnnarduchi at gmail.com (John Narduchi) Date: Thu, 29 Jun 2017 23:58:37 +0000 Subject: [Freeswitch-users] DTMF 2833 not working when switching to dynamic payload type 96 Message-ID: Having some trouble getting DTMF to work where an INVITE comes into FreeSWITCH offering a=rtpmap:101 telephone-event/16000. FreeSWITCH responds with a=rtpmap:96 telephone-event/8000 but then is unable to change the payload type to 96. This invite is a bit out of the ordinary as it does not specify telephone-event/8000. INVITE SDP *m=audio 63244 RTP/AVP 9 0 101* *b=AS:80* *a=rtpmap:101 telephone-event/16000* *a=fmtp:101 0-15* *a=ptime:20* *a=rtcp-xr:voip-metrics* *a=sendrecv* //fs logs after INVITE *2017-06-29 19:28:47.475013 [DEBUG] switch_core_media.c:4192 Set telephone-event payload to 101 at 16000* *2017-06-29 19:28:47.475013 [DEBUG] switch_core_media.c:4596 sofia/external/1231231234 at X.X.X.X Set 2833 dtmf send payload to 101 recv payload to 101* FreeSWITCH then replies with the following SDP. *m=audio 33734 RTP/AVP 0 96* *a=rtpmap:0 PCMU/8000* *a=rtpmap:96 telephone-event/8000* *a=fmtp:96 0-16* *a=silenceSupp:off - - - -* *a=ptime:20* When the digits come in to FS they are type 96, confirmed via tcpdump however FreeSWITCH doesn't recognize them and this entry shows up in the logs when digits are entered. *2017-06-29 19:29:02.975006 [WARNING] switch_core_media.c:2563 Could not change to payload type 96, ignoring...* *2017-06-29 19:29:02.975006 [DEBUG] switch_core_media.c:2540 alternate payload received (received 96, expecting 0)* Sofia.conf.xml and external.xml are vanilla. Have tested liberal-dtmf=true but no luck. Tested on version 1.6.10 and 1.6.18. Was hoping someone might have a setting to try or any suggestions? Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: From nathan.port at gmail.com Fri Jun 30 04:01:04 2017 From: nathan.port at gmail.com (Nate) Date: Fri, 30 Jun 2017 16:01:04 +1200 Subject: [Freeswitch-users] Dialplan: destination_number: regex question Message-ID: Good day/evening everyone, Apologies for not being able to figure this out on my own. I've been searching and trying for several days to get inbound/outbound working but have yet to see success. At this stage I need help determining a proper regex expression for handling New Zealand phone numbers. For instance, there are three different ways of expressing numbers here in NZ: International: +64 22 333 4444 non-local: 022 333 4444 local: 333 4444 A couple questions related to the SIP proxy: If the SIP proxy is located in Hong Kong, but the phone number is a New Zealand number, does the location of the proxy have any impact on the number of characters in the string for inbound/outbound calls? Again, apologies for the rudimentary nature of these questions, but having nearly exhausted all other options (docs, searches, IRC), I am now spending a large amount of time guessing and trial and error without any progress. Many thanks for any feedback at all. Nate -------------- next part -------------- An HTML attachment was scrubbed... URL: From shaun.stokes at itec-support.co.uk Fri Jun 30 11:04:04 2017 From: shaun.stokes at itec-support.co.uk (Shaun Stokes) Date: Fri, 30 Jun 2017 11:04:04 +0000 Subject: [Freeswitch-users] Limit In-Reply-To: <6FD2F8B5BB72834E9939AEDF9FB802A901E8678F9A@mbx-01.sysconfig.co.uk> References: <6FD2F8B5BB72834E9939AEDF9FB802A901E8678F9A@mbx-01.sysconfig.co.uk> Message-ID: <6FD2F8B5BB72834E9939AEDF9FB802A901E8678FDB@mbx-01.sysconfig.co.uk> Hi All, In our environment we restrict the number of current channels (using limit) per extension, however using attended transfer allows extensions to exceed the limit since leg a ends once the transfer completes which resets the limit to 0. I believe the solution is to apply the limit on leg b, however leg b is initiated via a bridge (with-in our outbound LUA script). How can we apply the limit to leg b, or is there a better solution? Thanks, Shaun [http://www.itec-support.co.uk/wp-content/uploads/2016/07/email_logo.jpg] Shaun Stokes - Infrastructure Analyst T : 01453 700713 E : shaun.stokes at itec-support.co.uk W : www.itec-support.co.uk Registered Address :- ITEC Support, Suite 2 Prospect House, Bath Road, Stroud, Gloucestershire GL5 3QF Company No. 06908001 CONFIDENTIALITY NOTICE This communication and the information it contains are intended for the person or organisation to which it is addressed. Its contents are confidential and may be protected in law. Unauthorised use, copying or disclosure of any of it may be unlawful. If you are not the intended recipient, please contact us immediately. The contents of any attachments in this e-mail may contain software viruses, which could damage your own computer system. While ITEC Support has taken every reasonable precaution to minimise this risk, we cannot accept liability for any damage which you sustain as a result of software viruses. You should carry out your own virus checking procedure before opening any attachment. ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL: From luis.daniel.lucio at gmail.com Fri Jun 30 11:44:48 2017 From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz) Date: Fri, 30 Jun 2017 07:44:48 -0400 Subject: [Freeswitch-users] Limit In-Reply-To: <6FD2F8B5BB72834E9939AEDF9FB802A901E8678FDB@mbx-01.sysconfig.co.uk> References: <6FD2F8B5BB72834E9939AEDF9FB802A901E8678F9A@mbx-01.sysconfig.co.uk> <6FD2F8B5BB72834E9939AEDF9FB802A901E8678FDB@mbx-01.sysconfig.co.uk> Message-ID: In your script, do a transfer instead a bridge. Let the dialplan deal with the bridge Le 30 juin 2017 7:05 AM, "Shaun Stokes" a écrit : > Hi All, > > > > In our environment we restrict the number of current channels (using > limit) per extension, however using attended transfer allows extensions to > exceed the limit since leg a ends once the transfer completes which resets > the limit to 0. > > > > I believe the solution is to apply the limit on leg b, however leg b is > initiated via a bridge (with-in our outbound LUA script). > > > > How can we apply the limit to leg b, or is there a better solution? > > > > Thanks, > > Shaun > Shaun Stokes - Infrastructure Analyst > T : 01453 700713 > E : shaun.stokes at itec-support.co.uk > W : www.itec-support.co.uk > Registered Address :- ITEC Support, Suite 2 Prospect House, Bath Road, > Stroud, Gloucestershire GL5 3QF > Company No. 06908001 > > CONFIDENTIALITY NOTICE > This communication and the information it contains are intended for the > person or organisation to which it is addressed. Its contents are > confidential and may be protected in law. Unauthorised use, copying or > disclosure of any of it may be unlawful. If you are not the intended > recipient, please contact us immediately. > The contents of any attachments in this e-mail may contain software > viruses, which could damage your own computer system. While ITEC Support > has taken every reasonable precaution to minimise this risk, we cannot > accept liability for any damage which you sustain as a result of software > viruses. You should carry out your own virus checking procedure before > opening any attachment. > > ______________________________________________________________________ > This message has been checked for all known viruses by MessageLabs Virus > Scanning Service. > ______________________________________________________________________ > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From krice at freeswitch.org Fri Jun 30 13:54:30 2017 From: krice at freeswitch.org (Ken Rice) Date: Fri, 30 Jun 2017 08:54:30 -0500 Subject: [Freeswitch-users] may be fix DNS? In-Reply-To: References: Message-ID: <16bc01d2f1a8$6604dc40$320e94c0$@freeswitch.org> Something appears to be wrong with the routing on your… the DNS is correct and we can ping google and various other places via IPv6 from that host. From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sergey Safarov Sent: Friday, June 30, 2017 2:40 AM To: FreeSWITCH Users Help Subject: [Freeswitch-users] may be fix DNS? [safarov at safarov ~]$ ping files.freeswitch.org PING files.freeswitch.org (2607:f348:1021::7 (2607:f348:1021::7)) 56 data bytes >From 2a05:ddc0::48 (2a05:ddc0::48) icmp_seq=1 Destination unreachable: No route >From 2a05:ddc0::48 (2a05:ddc0::48) icmp_seq=2 Destination unreachable: No route -------------- next part -------------- An HTML attachment was scrubbed... URL: From shaun.stokes at itec-support.co.uk Fri Jun 30 15:23:16 2017 From: shaun.stokes at itec-support.co.uk (Shaun Stokes) Date: Fri, 30 Jun 2017 15:23:16 +0000 Subject: [Freeswitch-users] Limit In-Reply-To: References: <6FD2F8B5BB72834E9939AEDF9FB802A901E8678F9A@mbx-01.sysconfig.co.uk> <6FD2F8B5BB72834E9939AEDF9FB802A901E8678FDB@mbx-01.sysconfig.co.uk> Message-ID: <6FD2F8B5BB72834E9939AEDF9FB802A901E86791EC@mbx-01.sysconfig.co.uk> I should also point out, the extensions are transferring to external PSTN numbers not local to FreeSWITCH. I’ll give this a try and will provide feedback, we’ll need a new dialplan to bridge the gateway. Thanks, Shaun From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Luis Daniel Lucio Quiroz Sent: 30 June 2017 12:45 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Limit In your script, do a transfer instead a bridge. Let the dialplan deal with the bridge Le 30 juin 2017 7:05 AM, "Shaun Stokes" > a écrit : Hi All, In our environment we restrict the number of current channels (using limit) per extension, however using attended transfer allows extensions to exceed the limit since leg a ends once the transfer completes which resets the limit to 0. I believe the solution is to apply the limit on leg b, however leg b is initiated via a bridge (with-in our outbound LUA script). How can we apply the limit to leg b, or is there a better solution? Thanks, Shaun [http://www.itec-support.co.uk/wp-content/uploads/2016/07/email_logo.jpg] Shaun Stokes - Infrastructure Analyst T : 01453 700713 E : shaun.stokes at itec-support.co.uk W : www.itec-support.co.uk Registered Address :- ITEC Support, Suite 2 Prospect House, Bath Road, Stroud, Gloucestershire GL5 3QF Company No. 06908001 CONFIDENTIALITY NOTICE This communication and the information it contains are intended for the person or organisation to which it is addressed. Its contents are confidential and may be protected in law. Unauthorised use, copying or disclosure of any of it may be unlawful. If you are not the intended recipient, please contact us immediately. The contents of any attachments in this e-mail may contain software viruses, which could damage your own computer system. While ITEC Support has taken every reasonable precaution to minimise this risk, we cannot accept liability for any damage which you sustain as a result of software viruses. You should carry out your own virus checking procedure before opening any attachment. ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ [http://www.itec-support.co.uk/wp-content/uploads/2016/07/email_logo.jpg] Shaun Stokes - Infrastructure Analyst T : 01453 700713 E : shaun.stokes at itec-support.co.uk W : www.itec-support.co.uk Registered Address :- ITEC Support, Suite 2 Prospect House, Bath Road, Stroud, Gloucestershire GL5 3QF Company No. 06908001 CONFIDENTIALITY NOTICE This communication and the information it contains are intended for the person or organisation to which it is addressed. Its contents are confidential and may be protected in law. Unauthorised use, copying or disclosure of any of it may be unlawful. If you are not the intended recipient, please contact us immediately. The contents of any attachments in this e-mail may contain software viruses, which could damage your own computer system. While ITEC Support has taken every reasonable precaution to minimise this risk, we cannot accept liability for any damage which you sustain as a result of software viruses. You should carry out your own virus checking procedure before opening any attachment. ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ -------------- next part -------------- An HTML attachment was scrubbed... URL: From lexxua at gmail.com Fri Jun 30 15:28:44 2017 From: lexxua at gmail.com (Volodymyr Fedorov) Date: Fri, 30 Jun 2017 17:28:44 +0200 Subject: [Freeswitch-users] Limit In-Reply-To: <6FD2F8B5BB72834E9939AEDF9FB802A901E86791EC@mbx-01.sysconfig.co.uk> References: <6FD2F8B5BB72834E9939AEDF9FB802A901E8678F9A@mbx-01.sysconfig.co.uk> <6FD2F8B5BB72834E9939AEDF9FB802A901E8678FDB@mbx-01.sysconfig.co.uk> <6FD2F8B5BB72834E9939AEDF9FB802A901E86791EC@mbx-01.sysconfig.co.uk> Message-ID: Hi Shaun, Is your problem looks like this ones ? https://freeswitch.org/jira/browse/FS-6778?filter=-2 Br, Volodymyr On Fri, Jun 30, 2017 at 5:23 PM, Shaun Stokes < shaun.stokes at itec-support.co.uk> wrote: > I should also point out, the extensions are transferring to external PSTN > numbers not local to FreeSWITCH. > > > > I’ll give this a try and will provide feedback, we’ll need a new dialplan > to bridge the gateway. > > > > Thanks, > > Shaun > > > > *From:* FreeSWITCH-users [mailto:freeswitch-users- > bounces at lists.freeswitch.org] *On Behalf Of *Luis Daniel Lucio Quiroz > *Sent:* 30 June 2017 12:45 > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Limit > > > > In your script, do a transfer instead a bridge. Let the dialplan deal with > the bridge > > > > Le 30 juin 2017 7:05 AM, "Shaun Stokes" > a écrit : > > Hi All, > > > > In our environment we restrict the number of current channels (using > limit) per extension, however using attended transfer allows extensions to > exceed the limit since leg a ends once the transfer completes which resets > the limit to 0. > > > > I believe the solution is to apply the limit on leg b, however leg b is > initiated via a bridge (with-in our outbound LUA script). > > > > How can we apply the limit to leg b, or is there a better solution? > > > > Thanks, > > Shaun > > Shaun Stokes - Infrastructure Analyst > > T : > > 01453 700713 > > E : > > shaun.stokes at itec-support.co.uk > > W : > > www.itec-support.co.uk > > Registered Address :- ITEC Support, Suite 2 Prospect House, Bath Road, > Stroud, Gloucestershire GL5 3QF > Company No. 06908001 > > > CONFIDENTIALITY NOTICE > This communication and the information it contains are intended for the > person or organisation to which it is addressed. Its contents are > confidential and may be protected in law. Unauthorised use, copying or > disclosure of any of it may be unlawful. If you are not the intended > recipient, please contact us immediately. > The contents of any attachments in this e-mail may contain software > viruses, which could damage your own computer system. While ITEC Support > has taken every reasonable precaution to minimise this risk, we cannot > accept liability for any damage which you sustain as a result of software > viruses. You should carry out your own virus checking procedure before > opening any attachment. > > > ______________________________________________________________________ > This message has been checked for all known viruses by MessageLabs Virus > Scanning Service. > ______________________________________________________________________ > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ______________________________________________________________________ > This message has been checked for all known viruses by MessageLabs Virus > Scanning Service. > ______________________________________________________________________ > Shaun Stokes - Infrastructure Analyst > T : 01453 700713 > E : shaun.stokes at itec-support.co.uk > W : www.itec-support.co.uk > Registered Address :- ITEC Support, Suite 2 Prospect House, Bath Road, > Stroud, Gloucestershire GL5 3QF > Company No. 06908001 > > CONFIDENTIALITY NOTICE > This communication and the information it contains are intended for the > person or organisation to which it is addressed. Its contents are > confidential and may be protected in law. Unauthorised use, copying or > disclosure of any of it may be unlawful. If you are not the intended > recipient, please contact us immediately. > The contents of any attachments in this e-mail may contain software > viruses, which could damage your own computer system. While ITEC Support > has taken every reasonable precaution to minimise this risk, we cannot > accept liability for any damage which you sustain as a result of software > viruses. You should carry out your own virus checking procedure before > opening any attachment. > > ______________________________________________________________________ > This message has been checked for all known viruses by MessageLabs Virus > Scanning Service. > ______________________________________________________________________ > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Best regards, Volodymyr -------------- next part -------------- An HTML attachment was scrubbed... URL: From wesleyakio at tuntscorp.com Fri Jun 30 16:21:25 2017 From: wesleyakio at tuntscorp.com (Wesley Akio) Date: Fri, 30 Jun 2017 16:21:25 +0000 Subject: [Freeswitch-users] may be fix DNS? In-Reply-To: <16bc01d2f1a8$6604dc40$320e94c0$@freeswitch.org> References: <16bc01d2f1a8$6604dc40$320e94c0$@freeswitch.org> Message-ID: I had more than a fair deal of issues resolving files.freeswitch.org... Last time I did a full recursion walkthrough all the way from the DNS root servers to prove there was an issue and its been dismissed just the same. On Fri, Jun 30, 2017 at 10:55 AM Ken Rice wrote: > Something appears to be wrong with the routing on your… the DNS is correct > and we can ping google and various other places via IPv6 from that host. > > > > *From:* FreeSWITCH-users [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Sergey > Safarov > *Sent:* Friday, June 30, 2017 2:40 AM > *To:* FreeSWITCH Users Help > *Subject:* [Freeswitch-users] may be fix DNS? > > > > [safarov at safarov ~]$ ping files.freeswitch.org > > PING files.freeswitch.org(2607:f348:1021::7 (2607:f348:1021::7)) 56 data > bytes > > From 2a05:ddc0::48 (2a05:ddc0::48) icmp_seq=1 Destination unreachable: No > route > > From 2a05:ddc0::48 (2a05:ddc0::48) icmp_seq=2 Destination unreachable: No > route > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- *Wesley Akio* Travessa da Lapa 96 - Sala 24 Centro – Curitiba – 80010-190 : +55 (41) 99950-0033 : +1 (305) 842-7095 : wesley_akio_ -------------- next part -------------- An HTML attachment was scrubbed... URL: From krice at freeswitch.org Fri Jun 30 17:11:28 2017 From: krice at freeswitch.org (Ken Rice) Date: Fri, 30 Jun 2017 12:11:28 -0500 Subject: [Freeswitch-users] may be fix DNS? In-Reply-To: References: <16bc01d2f1a8$6604dc40$320e94c0$@freeswitch.org> Message-ID: <1e3901d2f1c3$ea0a0e50$be1e2af0$@freeswitch.org> his diagnostic information clearly shows the correct ipv6 address being returned correctly. however , he can’t ping it, this would relate more to a routing issue than a DNS issue. it's also well known that there are islands of IPv6 out there due to several large providers who can’t play nice with each other. For example (unless this issue has been recently resolved) users single homed to Hurricane or Cogent cannot communicate with single homed users on the other network over IPv6 (even on ipv4 cogent doesn’t peer with HE although HE has requested it many times) From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Wesley Akio Sent: Friday, June 30, 2017 11:21 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] may be fix DNS? I had more than a fair deal of issues resolving files.freeswitch.org... Last time I did a full recursion walkthrough all the way from the DNS root servers to prove there was an issue and its been dismissed just the same. On Fri, Jun 30, 2017 at 10:55 AM Ken Rice > wrote: Something appears to be wrong with the routing on your… the DNS is correct and we can ping google and various other places via IPv6 from that host. From: FreeSWITCH-users [mailto:freeswitch-users-bounces at lists.freeswitch.org ] On Behalf Of Sergey Safarov Sent: Friday, June 30, 2017 2:40 AM To: FreeSWITCH Users Help > Subject: [Freeswitch-users] may be fix DNS? [safarov at safarov ~]$ ping files.freeswitch.org PING files.freeswitch.org (2607:f348:1021::7 (2607:f348:1021::7)) 56 data bytes >From 2a05:ddc0::48 (2a05:ddc0::48) icmp_seq=1 Destination unreachable: No route >From 2a05:ddc0::48 (2a05:ddc0::48) icmp_seq=2 Destination unreachable: No route _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at 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 at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Wesley Akio Travessa da Lapa 96 - Sala 24 Centro – Curitiba – 80010-190 : +55 (41) 99950-0033 : +1 (305) 842-7095 : wesley_akio_ -------------- next part -------------- An HTML attachment was scrubbed... URL: From rahul.ultimate at gmail.com Fri Jun 30 18:41:29 2017 From: rahul.ultimate at gmail.com (Rahul MathuR) Date: Sat, 1 Jul 2017 00:11:29 +0530 Subject: [Freeswitch-users] Switching profiles upon 3xx Redirection In-Reply-To: References: Message-ID: Hello guys, Thanks for a wonderful product ! I'm loving it. I am working on a situation where FS is installed on a centos7 having 2 ip address internal and public. On internal, I receive INVITE and create a b-leg and send to a Redirect server. This server sends 3xx and populates Contact header. When I receive it, I can see that external profile has already been set. And I can't change it to send it to another profile. Is there a way to do that ? My external profile listens on internal ip. I tried different options mentioned in confluence page but to no avail. Am I missing anything or trying to accomplish something impossible ? Thanks in anticipation. Rahul -------------- next part -------------- An HTML attachment was scrubbed... URL: From anthony.minessale at gmail.com Fri Jun 30 19:00:22 2017 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 30 Jun 2017 14:00:22 -0500 Subject: [Freeswitch-users] DTMF 2833 not working when switching to dynamic payload type 96 In-Reply-To: References: Message-ID: You would have to get a full log and file that to JIRA https://freeswitch.org/jira The mailing list is not for reporting bugs. On Thu, Jun 29, 2017 at 6:58 PM, John Narduchi wrote: > Having some trouble getting DTMF to work where an INVITE comes into > FreeSWITCH offering a=rtpmap:101 telephone-event/16000. FreeSWITCH responds > with a=rtpmap:96 telephone-event/8000 but then is unable to change the > payload type to 96. > > This invite is a bit out of the ordinary as it does not specify > telephone-event/8000. > > INVITE SDP > > *m=audio 63244 RTP/AVP 9 0 101* > *b=AS:80* > *a=rtpmap:101 telephone-event/16000* > *a=fmtp:101 0-15* > *a=ptime:20* > *a=rtcp-xr:voip-metrics* > *a=sendrecv* > > //fs logs after INVITE > *2017-06-29 19:28:47.475013 [DEBUG] switch_core_media.c:4192 Set > telephone-event payload to 101 at 16000* > *2017-06-29 19:28:47.475013 [DEBUG] switch_core_media.c:4596 > sofia/external/1231231234 at X.X.X.X Set 2833 dtmf send payload to 101 recv > payload to 101* > > > FreeSWITCH then replies with the following SDP. > > *m=audio 33734 RTP/AVP 0 96* > *a=rtpmap:0 PCMU/8000* > *a=rtpmap:96 telephone-event/8000* > *a=fmtp:96 0-16* > *a=silenceSupp:off - - - -* > *a=ptime:20* > > When the digits come in to FS they are type 96, confirmed via tcpdump > however FreeSWITCH doesn't recognize them and this entry shows up in the > logs when digits are entered. > > *2017-06-29 19:29:02.975006 [WARNING] switch_core_media.c:2563 Could not > change to payload type 96, ignoring...* > *2017-06-29 19:29:02.975006 [DEBUG] switch_core_media.c:2540 alternate > payload received (received 96, expecting 0)* > > Sofia.conf.xml and external.xml are vanilla. Have tested liberal-dtmf=true > but no luck. Tested on version 1.6.10 and 1.6.18. Was hoping someone might > have a setting to try or any suggestions? > > Thanks! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at 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 ♬ @anthmfs ♬ @FreeSWITCH ♬ ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ * ClueCon Weekly Development Call ☎ sip:888 at conference.freeswitch.org ☎ +19193869900 https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA -------------- next part -------------- An HTML attachment was scrubbed... URL: From anthony.minessale at gmail.com Fri Jun 30 19:11:29 2017 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 30 Jun 2017 14:11:29 -0500 Subject: [Freeswitch-users] Use FreeSWITCH to solve a problem and win a prize! Message-ID: This year at the ClueCon Coder Games there is a category for the most innovative use of FreeSWITCH to solve a problem. The prize is a bar top arcade. Get started now! Check out the other challenges and prizes on our web site: http://cluecon.com/coder-games.html ​ -- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ * ClueCon Weekly Development Call ☎ sip:888 at conference.freeswitch.org ☎ +19193869900 https://www.youtube.com/watch?v=oAxXgyx5jUw https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA -------------- next part -------------- An HTML attachment was scrubbed... URL: -------------- next part -------------- A non-text attachment was scrubbed... Name: il_570xN.1178461266_czuy.jpg Type: image/jpeg Size: 96298 bytes Desc: not available URL: From royj at yandex.ru Fri Jun 30 21:17:27 2017 From: royj at yandex.ru (royj at yandex.ru) Date: Sat, 01 Jul 2017 00:17:27 +0300 Subject: [Freeswitch-users] Dialplan: destination_number: regex question In-Reply-To: References: Message-ID: <3821491498857447@web5m.yandex.ru> An HTML attachment was scrubbed... URL: From kkothari157 at gmail.com Fri Jun 30 21:53:31 2017 From: kkothari157 at gmail.com (Ketan Kothari) Date: Sat, 1 Jul 2017 03:23:31 +0530 Subject: [Freeswitch-users] Audio issue in Amazon EC2 server In-Reply-To: References: Message-ID: Hello Giovanni, I have already set same in sip-profile but still getting same issue which mention. On Thu, Jun 29, 2017 at 6:46 PM, Giovanni Maruzzelli wrote: > > > On 29 June 2017 at 15:10, Ketan Kothari wrote: > >> Hi All, >> >> I have installed freeswitch on Amazon EC2 server and i followed *https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2 >> * >> >> Once we tried to call outbound call getting *no audio in both side* also >> FreeSWITCH passing my server local IP as RTP-IP to gateway side, >> > > > set your ext-rtp-ip and ext-sip-ip in all your sip_profiles to the > EXTERNAL (public) ip address of AWS instance > > -giovanni > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From gmaruzz at gmail.com Fri Jun 30 22:20:52 2017 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Sat, 1 Jul 2017 00:20:52 +0200 Subject: [Freeswitch-users] Audio issue in Amazon EC2 server In-Reply-To: References: Message-ID: On 30 June 2017 at 23:53, Ketan Kothari wrote: > Hello Giovanni, > > I have already set same in sip-profile but still getting same issue which > mention. > not possible. check again, you must edit "/usr/local/freeswitch/conf/sip_profiles/internal.xml" and "external.xml" files, then restart freeswitch Also, is a very strange private address 192.168.1.101 in AWS... Are you sure? -giovanni > > > > On Thu, Jun 29, 2017 at 6:46 PM, Giovanni Maruzzelli > wrote: > >> >> >> On 29 June 2017 at 15:10, Ketan Kothari wrote: >> >>> Hi All, >>> >>> I have installed freeswitch on Amazon EC2 server and i followed *https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2 >>> * >>> >>> Once we tried to call outbound call getting *no audio in both side* >>> also FreeSWITCH passing my server local IP as RTP-IP to gateway side, >>> >> >> >> set your ext-rtp-ip and ext-sip-ip in all your sip_profiles to the >> EXTERNAL (public) ip address of AWS instance >> >> -giovanni >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at 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 at 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 at 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 OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: From kkothari157 at gmail.com Fri Jun 30 22:28:41 2017 From: kkothari157 at gmail.com (Ketan Kothari) Date: Sat, 1 Jul 2017 03:58:41 +0530 Subject: [Freeswitch-users] Audio issue in Amazon EC2 server In-Reply-To: References: Message-ID: Hello Giovanni, Yeah i'm sure. I have checked in sip-profile ext-rtp-ip and ext-sip-ip both are set to public address but still while calling freeswitch sending local ip to gateway side. On Sat, Jul 1, 2017 at 3:50 AM, Giovanni Maruzzelli wrote: > > > On 30 June 2017 at 23:53, Ketan Kothari wrote: > >> Hello Giovanni, >> >> I have already set same in sip-profile but still getting same issue which >> mention. >> > > not possible. > check again, you must edit "/usr/local/freeswitch/conf/sip_profiles/internal.xml" > and "external.xml" files, then restart freeswitch > > > Also, is a very strange private address 192.168.1.101 in AWS... Are you > sure? > > -giovanni > > > >> >> >> >> On Thu, Jun 29, 2017 at 6:46 PM, Giovanni Maruzzelli >> wrote: >> >>> >>> >>> On 29 June 2017 at 15:10, Ketan Kothari wrote: >>> >>>> Hi All, >>>> >>>> I have installed freeswitch on Amazon EC2 server and i followed *https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2 >>>> * >>>> >>>> Once we tried to call outbound call getting *no audio in both side* >>>> also FreeSWITCH passing my server local IP as RTP-IP to gateway side, >>>> >>> >>> >>> set your ext-rtp-ip and ext-sip-ip in all your sip_profiles to the >>> EXTERNAL (public) ip address of AWS instance >>> >>> -giovanni >>> >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at 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 at 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 at 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 at 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 > OpenTelecom.IT > cell: +39 347 266 56 18 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: From gmaruzz at gmail.com Fri Jun 30 22:53:47 2017 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Sat, 1 Jul 2017 00:53:47 +0200 Subject: [Freeswitch-users] Audio issue in Amazon EC2 server In-Reply-To: References: Message-ID: On 1 July 2017 at 00:28, Ketan Kothari wrote: > Hello Giovanni, > > Yeah i'm sure. I have checked in sip-profile ext-rtp-ip and ext-sip-ip > both are set to public address but still while calling freeswitch sending > local ip to gateway side. > you edited "external.xml" as well? And restarted? can you post here the result of: "sofia status" and "sofia status profile external" from fs_cli? -giovanni > > On Sat, Jul 1, 2017 at 3:50 AM, Giovanni Maruzzelli > wrote: > >> >> >> On 30 June 2017 at 23:53, Ketan Kothari wrote: >> >>> Hello Giovanni, >>> >>> I have already set same in sip-profile but still getting same issue >>> which mention. >>> >> >> not possible. >> check again, you must edit "/usr/local/freeswitch/conf/sip_profiles/internal.xml" >> and "external.xml" files, then restart freeswitch >> >> >> Also, is a very strange private address 192.168.1.101 in AWS... Are you >> sure? >> >> -giovanni >> >> >> >>> >>> >>> >>> On Thu, Jun 29, 2017 at 6:46 PM, Giovanni Maruzzelli >>> wrote: >>> >>>> >>>> >>>> On 29 June 2017 at 15:10, Ketan Kothari wrote: >>>> >>>>> Hi All, >>>>> >>>>> I have installed freeswitch on Amazon EC2 server and i followed *https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2 >>>>> * >>>>> >>>>> Once we tried to call outbound call getting *no audio in both side* >>>>> also FreeSWITCH passing my server local IP as RTP-IP to gateway side, >>>>> >>>> >>>> >>>> set your ext-rtp-ip and ext-sip-ip in all your sip_profiles to the >>>> EXTERNAL (public) ip address of AWS instance >>>> >>>> -giovanni >>>> >>>> >>>> ____________________________________________________________ >>>> _____________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at 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 at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>> switch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at 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 at 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 >> OpenTelecom.IT >> cell: +39 347 266 56 18 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at 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 at 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 at 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 at 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 OpenTelecom.IT cell: +39 347 266 56 18 -------------- next part -------------- An HTML attachment was scrubbed... URL: From kkothari157 at gmail.com Fri Jun 30 23:33:46 2017 From: kkothari157 at gmail.com (Ketan Kothari) Date: Sat, 1 Jul 2017 05:03:46 +0530 Subject: [Freeswitch-users] Audio issue in Amazon EC2 server In-Reply-To: References: Message-ID: Hello Giovanni, Here is my "sofia status profile external" freeswitch@> sofia status profile external ================================================================================================= Name external Domain Name N/A Auto-NAT false DBName sofia_reg_external Pres Hosts 192.168.1.101 Dialplan XML Context default Challenge Realm auto_from RTP-IP 192.168.1.101 Ext-RTP-IP 128.22.54.32 SIP-IP 192.168.1.101 Ext-SIP-IP 128.22.54.32 URL sip:mod_sofia at 128.22.54.32:5060 BIND-URL sip:mod_sofia at 128.22.54.32:5060 ;maddr=192.168.1.101;transport=udp,tcp HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS IN PCMA,PCMU CODECS OUT PCMA,PCMU TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG false PROXY-MEDIA false ZRTP-PASSTHRU false AGGRESSIVENAT true CALLS-IN 2 FAILED-CALLS-IN 0 CALLS-OUT 2 FAILED-CALLS-OUT 0 REGISTRATIONS 0 =============== On Sat, Jul 1, 2017 at 4:23 AM, Giovanni Maruzzelli wrote: > > > On 1 July 2017 at 00:28, Ketan Kothari wrote: > >> Hello Giovanni, >> >> Yeah i'm sure. I have checked in sip-profile ext-rtp-ip and ext-sip-ip >> both are set to public address but still while calling freeswitch sending >> local ip to gateway side. >> > > you edited "external.xml" as well? And restarted? > > can you post here the result of: > > "sofia status" > > and > > "sofia status profile external" > > from fs_cli? > > -giovanni > > >> >> On Sat, Jul 1, 2017 at 3:50 AM, Giovanni Maruzzelli >> wrote: >> >>> >>> >>> On 30 June 2017 at 23:53, Ketan Kothari wrote: >>> >>>> Hello Giovanni, >>>> >>>> I have already set same in sip-profile but still getting same issue >>>> which mention. >>>> >>> >>> not possible. >>> check again, you must edit "/usr/local/freeswitch/conf/sip_profiles/internal.xml" >>> and "external.xml" files, then restart freeswitch >>> >>> >>> Also, is a very strange private address 192.168.1.101 in AWS... Are you >>> sure? >>> >>> -giovanni >>> >>> >>> >>>> >>>> >>>> >>>> On Thu, Jun 29, 2017 at 6:46 PM, Giovanni Maruzzelli >>> > wrote: >>>> >>>>> >>>>> >>>>> On 29 June 2017 at 15:10, Ketan Kothari wrote: >>>>> >>>>>> Hi All, >>>>>> >>>>>> I have installed freeswitch on Amazon EC2 server and i followed *https://freeswitch.org/confluence/display/FREESWITCH/Amazon+EC2 >>>>>> * >>>>>> >>>>>> Once we tried to call outbound call getting *no audio in both side* >>>>>> also FreeSWITCH passing my server local IP as RTP-IP to gateway side, >>>>>> >>>>> >>>>> >>>>> set your ext-rtp-ip and ext-sip-ip in all your sip_profiles to the >>>>> EXTERNAL (public) ip address of AWS instance >>>>> >>>>> -giovanni >>>>> >>>>> >>>>> ____________________________________________________________ >>>>> _____________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at 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 at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>>> switch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> ____________________________________________________________ >>>> _____________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at 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 at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/free >>>> switch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> >>> -- >>> >>> Sincerely, >>> >>> Giovanni Maruzzelli >>> OpenTelecom.IT >>> cell: +39 347 266 56 18 >>> >>> ____________________________________________________________ >>> _____________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at 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 at 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 at 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 at 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 > OpenTelecom.IT > cell: +39 347 266 56 18 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at 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 at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: