similar to: Can't comile bundled PJSIP on CentOS 7

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

Thanks. The configure run successfully. but I got the warning below.. checking for the ability of -lsrtp to be linked in a shared object... no configure: WARNING: *** configure: WARNING: *** libsrtp could not be linked as a shared object. configure: WARNING: *** Try compiling libsrtp manually. Configure libsrtp configure: WARNING: *** with ./configure CFLAGS=-fPIC --prefix=/usr configure:

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

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

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

Hi all, I just prepared a little tutorial on installing Asterisk 13 on CentOS 6.5 64-bit. See http://astrecipes.net/index.php?n=668 Hope you like. :) l. -- Loway - home of QueueMetrics - http://queuemetrics.com Try the WombatDialer auto-dialer @ http://wombatdialer.com

Hi list, I'm trying to create an asterisk 1.8 rpm with SRTP. I found mention of a libsrtp rpm, <http://qutecom.ipex.cz/RPMS/srtp-1.4.4-1.i386.rpm > in these instructions, <http://www.voip-info.org/wiki/view/Asterisk+SRTP> but it is unreachable (by me, anyway). The libSRTP source is here, <http://srtp.sourceforge.net/download.html>. Has this already been packaged for

On 7/20/17 8:47 AM, Marcelo Terres wrote: > Which version of Asterisk are you using? Are you compiling it with the > bundle pjproject ? > > --with-pjproject-bundled > > Regards, > > Marcelo H. Terres <mhterres at gmail.com <mailto:mhterres at gmail.com>> > IM: mhterres at jabber.mundoopensource.com.br > <mailto:mhterres at

On 7/19/17 2:37 AM, Marcelo Terres wrote: > This is the pjsip library. > > Is it possible to you to update pjsip for the latest version to test > if it solves the problem? > > On 18 Jul 2017 3:52 pm, "Carlos Chavez" <cursor at telecomab.mx > <mailto:cursor at telecomab.mx>> wrote: > > I am getting frequent segfaults on a new Asterisk

Apologies in advance if this has come up a thousand times before but is there any way to stop this error in 1.8 ? [ Feb 26 15:09:09] ERROR[6678] chan_sip.c: No SRTP module loaded, can't setup SRTP session. -- Thanks, Phil

Hi; It seems the SNOM Phones are requesting to have SRTP but I do not have the module res_srtp. I tried to compile it with asterisk 1.8, make menuselect, but I found that it can not be used (I am not able to select it) with the following details: Secure RTP SRTP Depends on: srtp E Can use: N/A Conflicts with: N/A So, how I can use it? What I have to do to know the reason for not being able to

On Sunday, January 26, 2020 3:58:31 PM CET Pete Biggs wrote: > > what does Centos 7 do with UPD packets having invalid checksums? > > By default I assume they are just dropped - that's what should happen. Hm that's what thought. > > Are such packets inevitably dropped? > > Applications can specifically disable checksum checking for the kernel > network stack

I tried installing the Asterisk 11 RHEL6 packages from packages.asterisk.org I followed this guide: https://wiki.asterisk.org/wiki/display/AST/Asterisk+Packages The SRTP support appears to be missing though. I notice libsrtp was not automatically installed as a dependency, and no srtp module exists under /usr/lib64/asterisk/modules Is it still necessary to do a source build every time SRTP is

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 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....): speexdsp-devel, gmime-devel, uriparser-devel, iksemel-devel, uw-imap-devel, hoard Then, I am running the following commands

I've been having some issues with srtp. so I checked which version of libsrtp I built asterisk 11.6 against. I'm on fedora 21, so libsrtp-1.4.4-13.20101004cvs.fc21.x86_64. From https://github.com/cisco/libsrtp it seems that latest release is 1.5.1, released a couple of weeks ago. I'm not a fan of the bleeding edge, but using a version 4+ years old seems strange even to me. But,

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

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

Hi folks, At least several endpoints (soft phone and desk phones) are supporting various 256 bit ciphers for SRTP these days. I *believe* libsrtp has been updated to allow this, and that only the code in Asterisk has not been been updated to allow these stronger ciphers. Would anyone with the know-how be willing/able to submit a patch ? Thank you, Kevin Long

The Asterisk Development Team has announced the release of Certified Asterisk 13.1-cert6. This release is available for immediate download at http://downloads.asterisk.org/pub/telephony/certified-asterisk The release of Certified Asterisk 13.1-cert6 resolves several issues reported by the community and would have not been possible without your participation. Thank you! The following are the