Displaying 20 results from an estimated 1000 matches similar to: "Problems linking asterisk against self-compiled openssl on CentOS 5"
2013 Aug 12
3
Asterisk 11.5.0
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
2012 Dec 20
2
asterisk 11 and no RTP
I have a CentOS 6.3 machine I installed Asterisk 11, worked fine...
I then tried to install on Cents 5.8, seemed to go fine... Then when I
placed a call I got this:
ast_rtp_instance_new: No RTP engine was found. Do you have one loaded?
Did a search and found issues with ARM and this problem but did not help
me, not using gtalk
or anything. Just call between two polycom phones on local network.
2013 May 02
1
Building Asterisk 11.4.0-rc1 with PJSIP 2.1
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
2020 Mar 13
2
Asterisk 16, 9.0 - res_rtp_asterisk compilation error
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:
2019 Dec 22
2
res_rtp_asterisk.so problem with minimal (ish) chan-sip based Asterisk
Hi,
For years I've been running a minimal (ish) SIP based Asterisk with
the modules based on chan-sip. For various reasons unrelated to
Asterisk the machine the latest incarnation of this configuration has
been updated to Debian Buster and thus to Asterisk 16. Since this
upgrade I have a dependency problem related to res_rtp_asterisk.so.
So the old config was:
[modules]
autoload=no
load
2009 Jun 02
0
Segfault on unload of chan_h323 in asterisk-1.4.25
When the support for h323plus was announced for Asterisk 1.4.25, I tried
to build this support in Asterisk. For this, I checked out the h323plus
CVS from SourceForge, which reported version 1.20.beta5, and also the
ptlib-2.4.2 source RPM from Fedora 10. I finally managed to build a
chan_h323 for Asterisk 1.4.25, which apparently loads correctly, but now
I see that I get a segfault whenever I
2020 Mar 13
1
Asterisk 16, 9.0 - res_rtp_asterisk compilation error
Le 13/03/2020 à 13:30, Joshua C. Colp a écrit :
> On Fri, Mar 13, 2020 at 9:27 AM Administrator <admin at tootai.net
> <mailto:admin at tootai.net>> wrote:
>
> 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 ->
2020 Sep 05
4
func_pjsip_aor.so: undefined symbol: ast_sip_location_retrieve_aor_contacts
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
2017 May 12
3
pjsip: asterisk can't decide which codec to use
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
2013 Oct 31
0
Trap invalide opcode error
Hello,
Using Ubuntu Server 12.04 and Asterisk 11.2.1.
I'm getting the following error when trying to start asterisk:
(Syslog) kernel: [ 1032.713864] asterisk[26918] trap invalid opcode ip:7fc272923076 sp:7fff928cf1b0 error:0 in codec_ilbc.so[7fc272921000+e000]
We were running Asterisk on a physical box, but moved it to a virtual environment. That went fine. Asterisk started normally and
2014 Oct 23
1
Auto video call hangup
Hi,
I use a simple scheme:
SIP video phone A (h264/Asterisk 1.8.11) <---IAX2 trunk----> SIP video
phone B (h264/Asterisk 11.7.0)
When calls from A to B and vice versa drop on pickup.
On B side:
[Oct 24 16:33:49] DEBUG[15590][C-00000012] res_rtp_asterisk.c: Setting the
marker bit due to a source update
[Oct 24 16:33:49] DEBUG[15590][C-00000012] res_rtp_asterisk.c: Setting the
marker bit
2019 Dec 12
2
asterisk pjsip webrtc rtp to private IP
examples of "interesting" information like ICE result and howto make
"minimal" configuration of pjproject.conf
i.e.
for debugging app_queue.so
core set debug 5 app_queue.so
for debugging RTP
core set debug 10 rtp_engine
core set debug 10 res_rtp_asterisk
rtp set debug on
logger.conf
rtp => debug,verbose(5)
so i mean
in
2014 Jan 30
1
Parking in Asterisk 12.0.0
Hi
I'm trying to get the rebuilt parking functionality to work in Asterisk
12.0.0.
In Asterisk 11.6.0 I managed to get a call to get parked by adding a
dynamic feature in features.conf for the DMTF sequence *# which called a
macro in extensions.conf, which then runned the ParkAndAnnounce
application, and the call got parked.
The syntax for ParkAndAnnounce I used was this (I don't
2019 Dec 12
2
asterisk pjsip webrtc rtp to private IP
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
2010 Nov 30
2
Error loading module 'chan_gtalk.so': libiksemel.so.3: cannot open shared object file: No such file or directory
Hello,
Can't get chan_gtalk.so module to load, neither res_jabber.so:
Asterisk*CLI> module load chan_gtalk.so
Unable to load module chan_gtalk.so
Command 'module load chan_gtalk.so ' failed.
[Dec 1 16:10:05] WARNING[2931]: loader.c:387 load_dynamic_module: Error
loading module 'chan_gtalk.so': libiksemel.so.3: cannot open shared object
file: No such file or directory
[Dec
2020 Feb 27
3
error compiling current git
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
2023 Apr 17
1
RTP address learning and timing problem
Hi Joshua,
Thank you for that. From the code it kind of looks like
STRICT_RTP_LEARN_TIMEOUT is a minimum, not a maximum:
if (!ast_sockaddr_isnull(&rtp->strict_rtp_address)
&& STRICT_RTP_LEARN_TIMEOUT < ast_tvdiff_ms(ast_tvnow(),
rtp->rtp_source_learn.start)) {
ast_verb(4, "%p -- Strict RTP learning complete - Locking on source address
%s\n",
Our call shows:
#
2023 Apr 18
1
RTP address learning and timing problem
I don't know in that specific output what happened. Your best course of
action is to add further logging or step through the logic with all of the
knowledge you have of the RTP streams to understand what is happening.
On Mon, Apr 17, 2023 at 8:52 PM David Cunningham <dcunningham at voisonics.com>
wrote:
> Hi Joshua,
>
> Thank you for that. From the code it kind of looks like
2020 Sep 24
2
Negotiates g729 but RTP contains g711
Hi,
I was able to use Unsniff to validate that the incoming 20 byte payloads of audio from the downstream IAX2 trunk was definitely G.729a whilst Asterisk 16.13.0 transcodes to G.711a unnecessarily. Media is confirmed as having been negotiated as g729 on all four streams. Nuance with this call is that it's from a WebRTC client which doesn't transmit any audio, could this be influencing
2020 Jan 15
1
Call disrupted...due to registration of third server?
We use Asterisk 14 to proxy calls between two servers, 10.0.0.192 to
10.0.0.228. But sometimes another of our servers becomes listed as a SIP
agent, even though the server's IP address isn't part of our sip.conf,
extensions.conf, nor any other config I know of. For example in the log
snippet below, the source server experienced an SDP renegotiation in the
middle of a call, and seemingly as