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... 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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... 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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. 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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 > 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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 > > _______________________________________________________________