similar to: How to master Asterisk version when cloning PJPROJECT ?

2014-08-14 12:03 GMT+02:00 Joshua Colp <jcolp at digium.com>: > Olivier wrote: >> >> Hello, > > > Kia ora, > > >> I'm giving Asterisk 12 and 13 my first try. >> >> When compiling PJPROJECT from source, as described in wiki page, I'm using >> this: >> git clone https://github.com/asterisk/pjproject pjproject >>

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

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

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

On Thu, Mar 31, 2016 at 10:30 AM, Brian Wilson <brian at wildsong.biz> wrote: > I think that I worked around the first issue (libasteriskssl) last night. > Should have gone to bed earlier instead. :-) > Will test your patches this morning. I build on a barebones Debian 8.x > virtual machine - not "older" unless you consider "stable" = "older". >

Until Asterisk 11 I could use sip.conf to set defaults for all phones (language, dtmf, vmexten, etc) and just leave many fields in the database as NULL. What would be the proper way to do this for Asterisk 13 and PJSIP? -- Telecomunicaciones Abiertas de M?xico S.A. de C.V. Carlos Ch?vez +52 (55)9116-91161

I'm working with a SIP provider to try and transition our sip connection with them to PJSIP. I thought I had transitioned the settings correctly, but whenever I attempt an Originate it never even tries to send any PJSIP messages. I'm currently running Asterisk 13.0.0. Anyone have any suggestions as to what I am doing wrong? The SIP provider says the latest version of Asterisk they have

Hello, A couple of questions in relation with Asterisk 12 on Debian Wheezy. 1. Can paquet libpjproject-dev (from wheezy-backport) be installed as the sole binary to add PJSIP stack to Asterisk 12 (compiled from source) ? 2. When compiling PJPROJECT from source (see https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+pjproject?src=search)), where should PJSIP .so files be located

Hello, I am currently trying to set up pjsip realtime and would like to have outbound-publish, inbound-publication, and asterisk-publication sorcery object types in ODBC realtime. Is that currently supported? I know that some object types are known working and others are not. I was curious what the status of those objects are. Thanks! Matt Hoskins | NPG Corp | Systems Architect

Hello Asterisk, Would someone be kind enough as to add the issue: grp_lock error when compiling against pjproject and solution: delete the rogue install in /usr/local/include To the WIKI page about installing pjsip. I tried to update the WIKI but don't seem to have a way to do it. I know it's not supposed to happen and I know what I did wrong, but it's hard to imagine

On 06/14/2017 at 10:17 PM, Joshua Colp wrote: > On Wed, Jun 14, 2017, at 05:09 PM, Michael Maier wrote: > > <snip> > >> >> I can now say, that asterisk / pjsip seams to work *mostly* as expected. >> Just one exception - and that's the package in question, which can't be >> seen in tcpdump. >> >> I extended the above patch by adding

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

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.

From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Richard Mudgett Sent: Wednesday, September 23, 2015 6:52 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] problems with PJSIP install on UBUNTU 14.04 On Wed, Sep 23, 2015 at 5:43 PM, Ryan, Travis <RyanT at

Greetings everyone, I am attempting to adjust the volume of a call using Set(VOLUME) in my extensions.conf file. I am finding that Set(VOLUME(TX)=x) and Set(VOLUME(RX)=y) have no discernable effect on my endpoints (Snom 300 IP phones). I have tried setting x and y to -30, -10, -3, -2, -1, 0, 1, 2, 3, 4, 10, and 100 and there appears to be no change on the phone volume. I can see that the

Hi All, I installed Astreisk 13beta with pjproject 2.3(2.2.1) from source code on CentOS7. -- https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+pjproject The installation is OK. But the connected SIP cilents (both Linphone on Windows7) cannot communicate. I hope your comment such as the testing for resolving the problem. My status is the following(1 and 2). Why 'Everyone

As you know, the ability to use a bundled version of pjproject was introduced with Asterisk 13.8.0. More info on the Asterisk Wiki <https://wiki.asterisk.org/wiki/display/AST/Building+and+Installing+pjproject#BuildingandInstallingpjproject-bundled> and in this email thread <http://lists.digium.com/pipermail/asterisk-users/2016-March/288685.html>. Since then I've fixed a few

Le 17/01/2020 à 11:54, Administrator a écrit : > > Le 15/01/2020 à 19:24, Administrator a écrit : >> Hi all, >> >> we face a strange behavior while connecting an Asterisk16 instance >> with PJSIP to 2 providers: we receive error 401 Unauthorized, both of >> them having Kamailio as front-end. With other providers -we don't >> know if they run

Thank you for the speedy reply. My originate string is something like the following where xxxxx is really the sip provider's supplied IP address 1234567890 is really the phone number I am dialing PJSIP/outbound.vitelity.net/1234567890 In the chan_sip based solution, it's... SIP/outbound.vitelity.net/1234567890 Have a great day! Dan -----Original Message----- From:

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]