similar to: pjsip startup errors when using "with-ssl" configure option

asterisk-16.13.0-rc2. Fedora 32 pjsip won't load because of undefined symbols: [Sep 4 14:19:25] ERROR[141137]: loader.c:2396 load_modules: Error loading module 'func_pjsip_aor.so': /usr/lib64/asterisk/modules/func_pjsip_aor.so: undefined symbol: ast_sip_location_retrieve_aor_contacts [Sep 4 14:19:25] ERROR[141137]: loader.c:2396 load_modules: Error loading module

On Thu, Feb 20, 2020 at 9:38 PM Patrick Wakano <pwakano at gmail.com> wrote: > Hello list, > Hope you are all doing well! > > I am facing a problem when compiling Asterisk 16.8.0 in a CentOS 6 box and > I wonder if someone can put some light on it. > Log history short, install_prereq fails to install the packages (not sure > how important they actually are....):

Hi Kevin! Thanks very much for your reply! Much appreciated! So I just have a remaining question from this, if the with-ssl is not mandatory to have the encryption support, what is it actually used for? Maybe it is some old flag which is not needed anymore and so can be ignored for now and possibly removed from the configure/makefile stuff for future releases? Kind regards, Patrick Wakano On

The current Asterisk 13 and master git branches have a new feature that will be included in 13.8.0: The ability to compile and run Asterisk with a bundled version of pjproject. ?? Why would you want to do this? Several reasons: - Predictability: When built with the ?bundled pjproject, you're always certain of the version you're running against, no matter where it's

Hello. After successful compilation 15.6.1 (bundled pjsip) and start asterisk i has error Symbol pjsip_tls_transport_start2 not found. /main/libasteriskpj.exports does not containg pjsip_tls_transport_start2 and pjsip_tls_transport_start. More: * All versions before (including 15.5) has not such error on this computer (ubuntu 18.04). * with 15.6.0, 15.6.1 has error on this computer

I am trying to enable full WebRTC support on asterisk-11.15 for installation on a CentOS 5 machine. Currently the distro cannot be upgraded to any later CentOS series. This CentOS series ships with openssl-0.9.8e, which lacks DTLS-SRTP support required for WebRTC. So I decided to build a parallel install of openssl. I chose the Fedora 21 package, openssl-1.0.1j, and built it on CentOS 5. The

Hello using pjproject 2.5.5 using asterisk-certified-13.8-cert1 Compiled pjproject 2.5.5 with : ./configure CFLAGS="-DNDEBUG -DPJ_HAS_IPV6=1" --prefix=/usr --libdir=/usr/lib64 --enable-shared --disable-video --disable-sound --disable-opencore-amr Compiled Asterisk 13 with ./configure --libdir=/usr/lib64 All pjproject modules are selectable in menuselect, so here no problem.

On Mon, Mar 7, 2016 at 2:53 PM, Jean-Denis Girard <jd.girard at sysnux.pf> wrote: > Hi, > > Le 07/03/2016 09:28, George Joseph a ?crit : > > PLEASE TRY THIS!! I'd love some feedback BEFORE 13.8.0 is released. > > I have tried GIT-master-ee5a944M on my Fedora 23 test server, and got: > > [pjproject] Unpacking /tmp/pjproject-2.4.5.tar.bz2 > [pjproject]

The Asterisk Development Team has announced the release of Asterisk 13.8.0. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/asterisk The release of Asterisk 13.8.0 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the issues resolved in this release: New

Ok so now I'm getting this when doing a make in asterisk... travis at pcimphone1:~/downloads/asterisk-13.5.0$ make [LD] chan_pjsip.o pjsip/dialplan_functions.o -> chan_pjsip.so /usr/bin/ld: /usr/local/lib/libpjsip-ua-x86_64-unknown-linux-gnu.a(sip_inv.o): relocation R_X86_64_32S against `.rodata' can not be used when making a shared object; recompile with -fPIC

On 16-08-16 04:38, George Joseph wrote: > > > On Mon, Aug 15, 2016 at 1:24 PM, Jonas Kellens > <jonas.kellens at telenet.be <mailto:jonas.kellens at telenet.be>> wrote: > > Hello > > using pjproject 2.5.5 > using asterisk-certified-13.8-cert1 > > > IIRC there were API changes in pjproject 2.5 that aren't accounted for > in

Hello, I'm setting up an Asterisk 13.13.1 cluster on two CentOS boxes. I followed this: cd /usr/src wget ... asterisk-13.13.1.tar.gz tar zxf asterisk-13.13.1.tar.gz cd asterisk-13.13.1 ASTERISK_CONFIGURE="--libdir=/usr/lib64 --prefix=/usr" ./configure ${ASTERISK_CONFIGURE} --with-pjproject-bundled make menuselect (shows res-srtp is available) make latest make command fails with

Jonas Kellens wrote: > Question : I noticed I received an error when installing pjproject > --with-external-srtp > > I do not seems to have the srtp capability. > (However I can easily install with "yum install libsrtp-devel") > > Can this have anything to do with the no-audio-problems that I'm having ?? WebRTC requires SRTP and Asterisk has to be built with it

On 16-08-16 17:45, George Joseph wrote: > > > On Tue, Aug 16, 2016 at 3:21 AM, Jonas Kellens > <jonas.kellens at telenet.be <mailto:jonas.kellens at telenet.be>> wrote: > > On 16-08-16 04:38, George Joseph wrote: >> >> >> On Mon, Aug 15, 2016 at 1:24 PM, Jonas Kellens >> <jonas.kellens at telenet.be <mailto:jonas.kellens at

Asterisk is on public IP (as described in the first email) i have 10 years experience in voip, 4 years webrtc in production. i know about ICE/STUN/DTLS-SRTP. yes, not every detail but the basic mechanism but i confess. i dont understand WHY Asterisk SOMETIMES switches destination IP in RTP. this is not only about ICE. its about RTP engine too which is Asterisk specific and Asterisk DEBUG is

On Fri, 11 Sep 2015, Denniston, Todd A CIV NAVSURFWARCENDIV Crane wrote: > If anyone has clues on how to build and add just a portion of > gst-plugins-bad into the existing EL OS set of gstreamer plugins, please > share (even URL pointers). > > [1] http://cgit.freedesktop.org/gstreamer/gst-plugins-bad/tree/ext/openh264 > [2] https://en.wikipedia.org/wiki/OpenH264 > [3]

I have the Asterisk book, it's enormous, the 4th edition as per http://www.asteriskdocs.org/. I'd like to do something like: http://www.voip-info.org/wiki/view/Asterisk+quickstart just to have two hardphones act as extensions and call each other. Is that a reasonable first task? I'm looking for the simplest litmus test for functionality possible. thanks, Thufir

On Fri, 11 Sep 2015, Robert Arkiletian wrote: > does anyone running C6 have h.264 check box on www.youtube.com/html5 ? > > if yes, can you please post the output of My understanding, which may be incorrect, was that the firefox in EL6 wasn't built with gstreamer support, so adding that library isn't sufficient. jh

hi all, can someone help? I have centos 6.8 trying to install asterisk 14.3.0-rc2 on it with options as stated below - ./configure --with-crypto --with-ssl --with-srtp=/usr/local/lib --with-jansson=/ --with-pjproject-bundled when I tried to run "make menuselect". i get the error below. Makefile:109: makeopts: No such file or directory **** **** The configure script must be executed

Hello setting "nat=no" or omitting "nat=" in peer definition does not help either. Still no audio. Why do you think this is a NAT issue ? IP and port information in SDP-body is correct. Kind regards. On 12-08-16 09:25, ????? ?????? wrote: > > Try delete nat from 770000wrtc settings ice should do the same > > > On Aug 11, 2016 10:00 PM, "Jonas