similar to: Asterisk 12 on Debian Wheezy

-----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

2014-08-13 19:06 GMT+02:00 Matthew Jordan <mjordan at digium.com>: > On Wed, Aug 13, 2014 at 12:01 PM, Olivier <oza.4h07 at gmail.com> wrote: >> After installing various packages, here is what I did: >> >> TDIR=/usr/src >> cd $TDIR >> PJOPTIONS="--prefix=/usr --enable-shared --disable-sound >> --disable-resample --disable-video

On 9/24/15, 8:10 AM, "asterisk-users-bounces at lists.digium.com on behalf of Joshua Colp" <asterisk-users-bounces at lists.digium.com on behalf of jcolp at digium.com> wrote: >On 15-09-24 08:54 AM, Ryan, Travis wrote: >> >> travis at pcimphone1:~/downloads/asterisk-13.5.0$ pkg-config --libs >> libpjproject -L/usr/local/lib -lpjsua2-x86_64-unknown-linux-gnu

> -----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:

On 9/24/15, 1:30 PM, "asterisk-users-bounces at lists.digium.com on behalf of Joshua Colp" <asterisk-users-bounces at lists.digium.com on behalf of jcolp at digium.com> wrote: >Ryan, Travis wrote: >>> >> That folder doesn?t have any libpj files in it. How do I make it find >>the >> real ones? > >You can delete libpjproject.pc from the places it

Hello, I'm having issues connecting throu PRI with the following error "Requested transfer capability: 0x00 - SPEECH" Below are the logs: == Using SIP RTP CoS mark 5 -- Executing [97052660 at voipphones:1] Set("SIP/4856-00000003", "CALLERID(num)=xxxxxxxxx") in new stack -- Executing [97052660 at voipphones:2] Dial("SIP/4856-00000003",

Hello list, how come on my Asterisk 1.6.2.11, I have no help available ?! asterisk*CLI> core show application Dial -= Info about application 'Dial' =- [Synopsis] Not available [Description] Not available [Syntax] Not available [Arguments] Not available [See Also] Not available Kind regards, Jonas. -------------- next part -------------- An HTML attachment was scrubbed...

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 would like to know the difference between encrypt the rtp and signaling between two asterisks, or putting an SBC in front of each Asterisk pbx. I'm trying to understand whether an SBC could fit an Asterisk deployment Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL:

Since it is Saturday evening (7PM EST) I am asking this on the list in case someone who knows sees it and can answer. Astricon is in my back yard for the first time, and I could hit you with a rock. I would always like to attend, and spoke at the 2007 Astricon in Phoenix but don't have the idle cycles. Question: Can I just go to Astricon and take the dCAP exam only? In and out? Cost? I

Just done a clean install of rc5 on a totally new machine and found the following:- /etc/init.d/asterisk start errors on line 109 - there is no 0 before $VERBOSITY as in the other lines. More interesting is that after make samples I have no iax2 available. Dave Cotton

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

I downloaded the latest 1.8 (27922) but got some segmentation faults. The first one was when it loaded cdr_odb, and so I changed menuselect not to compile that one, but the second one was when it tried to load chan_agent and so I stopped there to see if anyone else was seeing this. The agents.conf is all commented out except for [general] . Anyone know what is happening? Thanks. P.S. I deleted

How can I let asterisk immediately dials a trunk when off hook?

Hi We're using 1.8.23.1 on CentOS 5 and are trying to get accurate stats for queues. For a particular customer, when I run queue show <queue_name> I get the following numbers: <queue_name> has 0 calls (max unlimited) in 'ringall' strategy (17s holdtime, 94s talktime), W:0, C:175, A:44, SL:48.6% within 45s So from that data we look at 17s holdtime And assume that is the

Hello list, I need to know how to deal with a redundant network with only one asterisk server, which is receiving registrations from the end points on both of its ethernet ports. This means extension 201 is registering both from eth0 and from eth1. Is there a way/software which can act as a middle man between asterisk and the ethernet ports, and by default sends registrations to asterisk only

Hi All, I'm having an issue where Asterisk continuously sends out a GAZILLION "SIP NOTIFY" messages when a user has a voice message in their INBOX. This issue is only present when my SIP users and peers are configured from my ODBC backend (MySQL). A static configuration of users in sip.conf resolves this and everything works fine. I'd like to confirm the layout of the

<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.01 Transitional//EN"> <html> <head> <meta http-equiv="content-type" content="text/html; charset=ISO-8859-1"> </head> <body text="#000000" bgcolor="#ffffff"> <font size="+1">Does anyone have links to the most recent audiocodes

Evening, Has anyone seen a how-to on getting Asterisk to work with Google Talk and Google Voice? Thanks