- FreeSWITCH-users - FreeSWITCH Mailing Lists

Enabling new languages for Say
by Martin Boese 02 Jun '17

02 Jun '17

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 <phrases><macros> tags within the <language> 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

2 1

Multiple c params in SDP
by Grant Bagdasarian 02 Jun '17

02 Jun '17

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

2 1

testing 1234
by Ken Rice 02 Jun '17

02 Jun '17

This is just a test

8 11

FS 1.6 Video
by FERNANDO VILLARROEL 01 Jun '17

01 Jun '17

Dear All. I am follow https://freeswitch.org/confluence/display/FREESWITCH/Debian+8+Jessie But not works for me i received: root@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@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.

2 1

I get no Emails from the user list!
by Alexander Haugg 01 Jun '17

01 Jun '17

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!

2 1

FS to FS
by Rick Jarvis 01 Jun '17

01 Jun '17

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

2 2

WG: Turn configuration mod_sofia
by Alexander Haugg 01 Jun '17

01 Jun '17

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 "<param name="ext-rtp-ip" value=" stun:stun.freeswitch.org"/>" - ICE configuration could be? "<param name="ext-rtp-ip" value=" { 'stun:stun.freeswitch.org', 'turn:turn.myserver.com;username=xxx;password=123'}"/>" 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: <a9e067a18e104afbb53e207712467baf(a)c4b.de<mailto:a9e067a18e104afbb53e207712467baf@c4b.de>> References: <a9e067a18e104afbb53e207712467baf(a)c4b.de<mailto:a9e067a18e104afbb53e207712467baf@c4b.de>> Message-ID: <CAEJMVkDy5e2COC8cH=xV=ozM0gepBS3cQ_YxUPQk4twMoaSVmg(a)mail.gmail.com<mailto:CAEJMVkDy5e2COC8cH=xV=ozM0gepBS3cQ_YxUPQk4twMoaSVmg@mail.gmail.com>> 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(a)lists.freeswitch.org' <freeswitch-users(a)lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org>> Betreff: Turn configuration mod_sofia Hi all, the stun configuration for the mod_sofia profile is very easy "<param name="ext-rtp-ip" value=" stun:stun.freeswitch.org"/>" 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

1 0

Embedding FreeSWITCH
by Paul Mateer 01 Jun '17

01 Jun '17

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

1 0

replace b-leg in a bridge on 302 moved temporarily
by Dist Lists 01 Jun '17

01 Jun '17

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?

1 0

originated_legs
by Tihomir Culjaga 01 Jun '17

01 Jun '17

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=<ext1_uuid>}user/1002 & park() ... and eventually i do uuid_bridge <in_call_uuid> <ext1_uuid> 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.

1 1