Displaying 20 results from an estimated 10000 matches similar to: "Segmentation fault after upgrading from asterisk-10.5.0 to asterisk-11.1.2"
2006 Apr 20
3
Asterisk Won't start after SVN Trunk Update
Hi:
I deleted old modules in /usr/lib/asterisk/modules
before make install. I built zaptel and libpri before
asterisk. Modprobe zaptel and modprobe -v wctdm
executed witiout complaint. Starting asterisk
produced the output below with several warnings and a
failure. Can someone help, please. I double-spaced
the warnings in the text below. The first warning is
about music on hold because it
2013 Jul 12
1
Using PauseMonitor with MixMonitor
Hi
I'm using asterisk 1.8 on CentOS 5
I'm initiating call recordings with MixMonitor and trying to pause them
with the features.conf.
Whenever I try to pause the recording the call dies. Is PauseMonitor
incompatible with MixMonitor?
Here are some key log excerpts
features reload
== Parsing '/etc/asterisk/features.conf': == Found
== Registered Feature
2012 Jan 18
1
Compile error 1.8.8.1
Hi,
While compiling 1.8.8.1, I met the following error:
[AR] hash/hash.o hash/hash_bigkey.o hash/hash_buf.o hash/hash_func.o
hash/hash_log2.o hash/hash_page.o hash/ndbm.o btree/bt_close.o
btree/bt_conv.o btree/bt_debug.o btree/bt_delete.o btree/bt_get.o
btree/bt_open.o btree/bt_overflow.o btree/bt_page.o btree/bt_put.o
btree/bt_search.o btree/bt_seq.o btree/bt_split.o btree/bt_utils.o db/db.o
2010 Apr 29
1
Starting call recording using a dynamic feature to call a macro
I have got call recording working on our 1.4.30 asterisk box together
with a recording pause ability and being able to play different audio to
each party at the start and end of the pause. This all works perfectly
but one wish is to have the audio files have a beep or something in them
so when you listen later you can tell where the audio was paused.
So I changed things around so that instead
2016 Dec 10
6
failing to start asterisk on centos7
ive installed asterisk but below is what am getting proces gets
killed.please help
[root at localhost sounds]# asterisk -vvvvc
Asterisk 13.13.1, Copyright (C) 1999 - 2014, Digium, Inc. and others.
Created by Mark Spencer <markster at digium.com>
Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for
details.
This is free software, with components licensed under
2011 Mar 07
1
[1.8.3] Error compiling Asterisk: __sync_fetch_and_add
Hello all,
mmm a bit embarrassing about not having a clue as to why we're getting this
error on make of 1.8.3
[AR] hash/hash.o hash/hash_bigkey.o hash/hash_buf.o hash/hash_func.o
hash/hash_log2.o hash/hash_page.o hash/ndbm.o btree/bt_close.o
btree/bt_conv.o btree/bt_debug.o btree/bt_delete.o btree/bt_get.o
btree/bt_open.o btree/bt_overflow.o btree/bt_page.o btree/bt_put.o
btree/bt_search.o
2010 Oct 24
5
Integrating Asterisk 1.8 with Google Talk and Google Voice
Evening,
Has anyone seen a how-to on getting Asterisk to work with Google Talk
and Google Voice?
Thanks
2015 May 21
4
PJSIP CCSS
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Le 21/05/2015 00:16, Joshua Colp a ?crit :
> If CCSS is needed then the only option is to use chan_sip. The
> chan_pjsip module does not implement CCSS in any way.
Is CCSS support planned for PJSIP? chan_sip is in "extended" state in
asterisk-13, so chan_pjsip should be preferred for new installations, ri
ght?
Thanks,
- --
2010 Dec 22
5
* 1.8: cannot load g729 free codec (on 1.4 it worked!)
pbx18*CLI> module load codec_g729-ast14-gcc4-glibc-pentium3.so
Unable to load module codec_g729-ast14-gcc4-glibc-pentium3.so
Command 'module load codec_g729-ast14-gcc4-glibc-pentium3.so ' failed.
[Dec 22 15:52:45] WARNING[4491]: loader.c:757 inspect_module: Module
'codec_g729-ast14-gcc4-glibc-pentium3.so' does not provide a license key.
[Dec 22 15:52:45] WARNING[4491]:
2009 May 11
1
PauseMonitor() Hanging Up Call
Hi All,
I'm at the end of my tether here and would really appreciate some help.
I'm trying to implement DTMF based pause/resume of call recording. I'm
using Asterisk 1.4.22.1.
Here's the scenario:
The caller (SIP or ISDN, doesn't matter) dials into the asterisk which
executes the following code:
exten => _X.,1,Monitor(wav,${CALLDIR}${UNIQUEID},mb)
2015 May 21
2
PJSIP CCSS
-----BEGIN PGP SIGNED MESSAGE-----
Hash: SHA1
Hi list,
It looks like Call Completion Supplementary Services is not available
for PJSIP channels, am I right? Is there another solution?
Thanks,
- --
Jean-Denis Girard
SysNux Syst?mes Linux en Polyn?sie fran?aise
http://www.sysnux.pf/ T?l: +689 40.50.10.40 / GSM: +689 87.79.75.27
-----BEGIN PGP SIGNATURE-----
2015 May 21
1
PJSIP CCSS
Ludovic Gasc wrote:
> 2015-05-21 17:59 GMT+02:00 Jean-Denis Girard <jd.girard at sysnux.pf
> <mailto:jd.girard at sysnux.pf>>:
>
> -----BEGIN PGP SIGNED MESSAGE-----
> Hash: SHA1
>
> Le 21/05/2015 00:16, Joshua Colp a ?crit :
> > If CCSS is needed then the only option is to use chan_sip. The
> > chan_pjsip module does not implement
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
2009 Jan 22
1
Zap connection problem
Greetings all,
I'm trying to connect to an AT&T teleconference, but the
call is never marked as ANSWERED by asterisk and therefore won't bridge and
continue. The only work-around I've come up with so far is to dial like
this:
Exten => 744,1,Dial(Zap/g1,,p)
The "private" mode keeps the line open without trying to do a bridge, but
requires the
2011 Nov 28
1
centos asterisk 1.8 rpms: chan_gtalk and res_jabber missing?
Hi All,
While I'm certainly comfortable compiling from sources, I'm trying to do an
rpm only asterisk install on CentOS 5.7. I'm using the asterisk
repositories and I installed all the asterisk18 rpms, but find that
chan_gtalk and res_jabber are missing.
Is there a separate rpm that includes support for gtalk?
Thanks in advance.
-Gaurav
-------------- next part --------------
An
2008 Feb 22
1
FW: jabber
Hi all,
Do some one experiencing running jabber applications (jabberstatus...) in
asterisk? I do experinced Asterisk 1.4.18 and wish to start it, however I
got such result.
IBM*CLI> help jabber
No such command 'jabber'.
IBM*CLI> help jabberstatus
No such command 'jabberstatus'.
Any one can help me on this, or may be I miss out somethings that cause
jabber applications
2007 Feb 03
2
Google Talk without gmail accout?
Hello all,
I am having trouble getting gtalk to work with my account which is not
using a gmail.com email address. When I do this there an error from the
Jabber module:
[Feb 3 20:51:17] ERROR[6286]: res_jabber.c:573 aji_act_hook: JABBER:
Node Error
[Feb 3 20:51:17] WARNING[6286]: res_jabber.c:1495 aji_recv_loop:
JABBER: Got hook event.
JABBER: gtalk_account INCOMING:
2007 Jul 19
2
Gtalk/Jabber connect issues in 1.4.8
I've included my jabber.conf below. I'm betting the following errors:
[Jul 18 21:05:22] ERROR[28166]: res_jabber.c:609 aji_act_hook: JABBER:
Node Error
[Jul 18 21:05:22] WARNING[28166]: res_jabber.c:1537 aji_recv_loop:
JABBER: Got hook event.
jabber test
[Jul 18 21:04:16] WARNING[32691]: res_jabber.c:1421 ast_aji_send:
JABBER: Not connected can't send
User: bferrell at gtalk.com
2012 Feb 02
1
T38 faxing - UDPTL creation failed
Hello guys.
When I am trying to send fax through T38 to linksys SPA (properly
configured etc. - I have tried it with other systems), I'm getting error
and fax is not delivered.
I'm getting this errors in asterisk.log:
WARNING[687] udptl.c: No UDPTL ports remaining
ERROR[687] chan_sip.c: UDPTL creation failed
WARNING[687] udptl.c: No UDPTL ports remaining
then, couple lines down:
2017 Jun 16
3
asterisk 13.16 / pjsip / t.38: res_pjsip_t38.c:207 t38_automatic_reject: Automatically rejecting T.38 request on channel 'PJSIP/91-00000007'
On Fri, Jun 16, 2017, at 10:49 AM, Michael Maier wrote:
<snip>
>
> t38modem and asterisk are using
>
> m=image 35622 udptl t38
> ^^^^^
>
> Provider uses
>
> m=image 35622 UDPTL t38
> ^^^^^
>
> Could this be a problem? If I'm sending internal only, it's always
> lowercase.
Looking at the tests we have we