similar to: Segfault in libpjnath.so.2 though PJSIP not present in dialplan

Le lun. 26 août 2019 à 12:07, Joshua C. Colp <jcolp at digium.com> a écrit : > ... > > libpjnath is the ICE/STUN/TURN library which is used by res_rtp_asterisk > for that functionality. If you're using WebRTC or ICE/STUN/TURN, then you > would be using that library. > Yes, I'm using ICE/STUN/TURN. That explains libpjnath usage. Thank you sharing this here. Now

Hello, I'm working on building Asterisk 11.4.0-rc1 with pjproject 2.1 instead of 2.0 due to a crashing issue resulting from ICE. https://issues.asterisk.org/jira/browse/ASTERISK-21696 Currently, I'm systematically going through each Makefile in every directory in pjproject and changing the paths that exist in the pjproject 2.0 included with Asterisk, so that I can successfully build

> -----Original Message----- > From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users- > bounces at lists.digium.com] On Behalf Of Joshua Colp > Sent: Wednesday, September 23, 2015 9:39 AM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: Re: [asterisk-users] problems with PJSIP install on UBUNTU > 14.04 > > Ryan, Travis wrote:

-----Original Message----- From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Joshua Colp Sent: Thursday, September 24, 2015 6:23 AM To: asterisk-users at lists.digium.com Subject: Re: [asterisk-users] problems with PJSIP install on UBUNTU 14.04 On 15-09-23 10:25 PM, Ryan, Travis wrote: > Ok I did all that and it's still

Ok I did all that and it's still crashing. I did find some other areas I think that shouldn't have had any of those files, so I thought it would work, because I got rid of ALL of them per your instructions and completely reconfigured and compiled both pjsip and then asterisk. Attached is the new backtrace... -----Original Message----- From: asterisk-users-bounces at lists.digium.com

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

I've built PJSIP a few months ago on a server that was 12.04 and can't remember how I got past this same issue. I've looked at the links I'll put below and the comments section where others had the issue, but those tips aren't helping either. Basically everything seems to compile and install correctly, but then the "ldconfig -p | grep pj" doesn't show anything.

Hello, I have implemented a pass to count total number of instructions in LLVM 8.0. Its source code is attached here. The pass runs fine with LLVM 4.0. But with LLVM 8.0 following error is shown; $ $LLVMopt_BIN/opt -load $LLVM_SO/LLVMStatic-Info.so -one output-simple.bc WARNING: You're attempting to print out a bitcode file. This is inadvisable as it may cause display problems. If you REALLY

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

Linearized array addresses are an issue in loop nest transformation. To alleviate the same, a multidimensional array indexing intrinsics have been proposed recently in the llvmp-dev mailing list [1]. From the mailing thread, it looks like there is a consensus on using intrinsics for communicating dimensions [2]. While working with our own loop transformation framework, we did a similar work on

hi, i have following topology PSTN - Asterisk ---- internet -----  router - jssip client (wss) Asterisk 13.29.1 on public IP, chan_pjsip for wss, chan_sip/udp for SIP connection to PSTN router - public IP/private IP (NAT) jssip client - private IP - sip over websocket to Asterisk PJSIP ~30% of calls has problem with no audio. reason is that Asterisk is sending RTP to private IP of jssip

Hello, What does it mean when "rtp set debug ip" shows RTP packets that have been send, but there is no audio ? There was no audio on my call in both directions, but "rtp set debug" shows that there were RTP packets send. There is no firewall active on my Asterisk server : [root at sip asterisk]# /sbin/service iptables status iptables: Firewall not running. Kind

I think DAGToDAG is too late because the build_vector has already been turned into a constant pool load by then so it’s a little difficult to get back. Maybe we can delay it to !DCI.isBeforeLegalizeOps()? That would at least let the first DAG combine and the post type legalization DAG combine see the add, 1. +Sanjay as well From: Amaury Séchet <deadalnix at gmail.com> Sent: Monday, August

Hi all, As you knwo already, I'm trying to change DAGCombiner so that it process the nodes in topological order. Doing so is not difficult per se, but this creates various improvements and regression to the existing test suite. I'd like to work through as many of the regressions as possible ahead of time. One source of such regressions is combineIncDecVector in the X86 backend. It

Hello, 2 asterisk servers 16.8.0 version running on Debian 10.3 On one of them, I can't compile asterisk having error    [CC] res_rtp_asterisk.c -> res_rtp_asterisk.o res_rtp_asterisk.c:2674:3: error: ‘pj_ice_sess_cb’ {aka ‘struct pj_ice_sess_cb’} has no member named ‘on_valid_pair’   .on_valid_pair = ast_rtp_on_valid_pair,    ^~~~~~~~~~~~~ res_rtp_asterisk.c:2674:19: warning:

Hello! I'm facing completely choppy sound. The wireshark trace shows, that there are a lot of codec changes without any trigger (means no options or reinvite or any other package). Background: The call is initiated by asterisk and is received by the same asterisk conference room via Phone extension -> asterisk -> provider A -> provider B -> asterisk. Asterisk initially sends

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

Hi, compiling the current git version on Centos 7 gives me: [CC] res_statsd.c -> res_statsd.o res_rtp_asterisk.c:2669:2: error: unknown field ‘on_valid_pair’ specified in initializer .on_valid_pair = ast_rtp_on_valid_pair, ^ res_rtp_asterisk.c:2669:2: warning: initialization from incompatible pointer type [enabled by default] res_rtp_asterisk.c:2669:2: warning: (near initialization

I have been using 11.4.0 for some time. All was fine. I downloaded 11.5, extracted, run ./configure, make, make install. I got a message about res_rtp_asterisk.so was not compiled in the 11.5 Sure enough I have rss_rtp_asterisk.c but not .o file and no .so file. I then looked in the config.log and nothing is in there about res_rtp_asterisk What's up? jerry -------------- next part

with wireshark i need decrypt traffic every call which is time consuming. get debug from pjnat through asterisk is not possible because of technical reasons or nobody did it? in my case its strange that ice candidates are the same good call v=0 o=- 3669976329745317845 2 IN IP4 127.0.0.1 s=- t=0 0 a=msid-semantic: WMS EoNIdKcMZvWBLULGqGPJTDe12ujjFEemeapo m=audio 52421 RTP/SAVPF 8 0 101 c=IN