Displaying 20 results from an estimated 4000 matches similar to: "issues.asterisk.org"
2011 Jun 01
1
Migration from Mantis to JIRA
Greetings,
A few weeks ago I posted a message about the upcoming migration from
Mantis to JIRA for issues.asterisk.org [1]. A lot of testing has been
done and all known issues have been resolved. We have scheduled the
migration for Sunday, June 5th. The issue tracker will be down most of
the day as the migration takes place. Once the migration is complete,
the issue tracker will be:
2012 Aug 27
3
Asterisk community services - Old Mantis instance to be shutdown on Aug 28th, 2012
On June 5, 2011, we migrated from Mantis to Jira as the issue tracker
for Asterisk [1]. We temporarily left Mantis running in read-only mode
to smooth the transition. At 15 months, temporary has turned into
semi-permanent. As a part of other infrastructure changes we are making
to the community services, we will finally shut down Mantis for good.
We will update our DNS servers on the morning of
2012 Aug 27
3
Asterisk community services - Old Mantis instance to be shutdown on Aug 28th, 2012
On June 5, 2011, we migrated from Mantis to Jira as the issue tracker
for Asterisk [1]. We temporarily left Mantis running in read-only mode
to smooth the transition. At 15 months, temporary has turned into
semi-permanent. As a part of other infrastructure changes we are making
to the community services, we will finally shut down Mantis for good.
We will update our DNS servers on the morning of
2013 Feb 12
1
asterisk 11 AGI
I recently upgraded to asterisk 11 from 1.8.
I had VXML working via AGI in 1.8 - from extensions.conf:
[VXML]
exten => s,1,Answer
exten => s,n,Set(ENCODED=${URIENCODE(${ARG1})})
exten => s,n,AGI(agi://localhost/url=${ENCODED})
exten => s,n,Hangup
Using asterisk 11 on the same host with the same config in extensions.conf:
-- Executing [s at VXML:1]
2010 Nov 13
2
asterisk 1.8 fax woes
I upgraded from a perfectly working 1.6.2 asterisk installation
(including fax via app_fax_digium) to 1.8.0 this evening.
All my custom modules (including swift <thanks darren!>) are working
fine except for fax.
When a caller connects, asterisk switches to the fax context and hangs
up the call.
i've captured with:
core set verbose 10
core set debug 10
fax set debug on
sip
2010 Nov 15
2
SIP calls destroyed after 1:20
After upgrading to Asterisk 1.8.0, I am finding that my outbound SIP
calls are being destroyed after 1 minute and 20 seconds (80 seconds).
Asterisk is sending a BYE message - I have no idea why.
http://jeremy.kister.net/tmp/20101115/sip.txt for a sip debug.
can anyone suggest how i can further deal with this?
--
Jeremy Kister
http://jeremy.kister.net./
2009 Jul 06
3
How to make big MySQL database more diffable/rsyncable? (aka rsyncing big files)
Hello group,
I'm having a very hard time rsyncing efficiently a MySQL database which
contains very large binary blobs.
(Actually, it's the database of Mantis bug tracker
[http://www.mantisbt.org/], with file attachments stored directly in the
table rows. I know it's a bad idea from many other reasons, but let's
say it was given to me as such.)
First, I was dumping the
2013 Oct 04
1
OT: Asterisk loses Oprah on live TV
just thought this was cute enough to pass along,
https://www.youtube.com/watch?feature=player_detailpage&v=GLwct15X_3g#t=135
--
Jeremy Kister
http://jeremy.kister.net./
2011 Jun 01
0
Migration from Mantis to JIRA
Greetings,
A few weeks ago I posted a message about the upcoming migration from
Mantis to JIRA for issues.asterisk.org [1]. A lot of testing has been
done and all known issues have been resolved. We have scheduled the
migration for Sunday, June 5th. The issue tracker will be down most of
the day as the migration takes place. Once the migration is complete,
the issue tracker will be:
2011 Mar 29
1
wrong from URI in options message
I recently configured a SIP peer which i must specify my fromuser as my
ten digit "DID". I send calls to this peer, but whenever Asterisk sends
an options message, the fromuser is "asterisk".
Is this a bug? Or is there some other config I must make ?
register = 2155551941:123456 at 10.0.138.226/2155551941~600
[peer](!)
type=peer
context=inbound
qualify=yes
2010 Nov 04
2
useless mpg123 processes hanging around
Running Asterisk 1.6.2.11 on debian 5.0.6 with mpg123 1.4.3
when i start asterisk, i immediately see two mpg123 processes spawned
which sit there forever. I can't imagine it's normal behavior, but if
it is, please explain :)
# /etc/init.d/asterisk stop
stopping asterisk.
#[...]
# /etc/init.d/asterisk start
starting asterisk.
# psg aster
root 14573 1 0 16:29 pts/2 00:00:00
2010 Jan 10
2
app_swift 1.6.2 DTMF issue
With app_swift 1.6.2 + asterisk 1.6.1.12, I've found that if you
enter DTMF during cepstral playback, the first digit of ${SWIFT_DTMF}
is [un]set in an odd way.
for example consider:
999,1,Swift(some long message that you dont want to wait for|5000|5)
999,n,NoOp(DTMF: ${SWIFT_DTMF})
if while I am listening to the playback, i interrupt and dial:
- "12345", SWIFT_DTMF is set to
2010 Aug 29
1
evil disconnect of call with cisco 1760
I have asterisk 1.6.2.11 talking to a cisco 1760 running
ipvoicek9-mz.124-25b.
whenever a call goes through the 1760's FXO or FXS (in or out) there is
a 915 second maximum call time due to asterisk hanging up the call
because of a "critical packet" being missed.
I read doc/sip-retransmit.txt and I don't see anything there that is
helpful to my situation - the asterisk box is
2013 Oct 10
1
asterisk 11.6 nat problem
using asterisk 11.6.0-rc1 i just converted my "nat=yes" to
"nat=auto_force_rport,auto_comedia"
I have my asterisk box on the same subnet as a cisco 1760 (vgw1).
a few times per day, Asterisk thinks vgw1 is dead (by qualify/options).
A 'sip reload' always fixes the problem.
i left 'sip set debug peer vgw1' on the console. but i dont see what's
2012 Jan 23
1
ConfBridge details
running Asterisk 1.8.9.0-rc2, what are the ways to interface with
ConfBridge ?
I see the CLI command 'confbridge' documented for asterisk 10, but i
dont see how to interface with confbridge on 1.8
What I'm trying to do is keep track of conferences that are used.
I tried something like the below, but not only does Confbridge not
return, but i'd need something that erases the
2011 Apr 15
3
sip error logging
I recently noticed that asterisk is not logging unknown sip connections.
I'm not sure if I've broken something or if asterisk itself has been
broken.
the last entry I have is:
/var/log/asterisk/messages.9:[Feb 10 23:58:48] NOTICE[9868] chan_sip.c:
Registration from '<sip:22942 at 10.0.0.3>' failed for '10.0.0.228:5060' -
No matching peer found
my logger.conf
2015 Apr 29
2
PJSIP - sessions-timers support not working on 13.X
Hi Josua, Sorry for writing wrong the parameter but i just copy paste the examples on pjsip.conf it wasn?t a "typo? error of timers parameters, i have an error on global tag and can?t load the timers
I was getting this :
[Apr 29 17:21:49] WARNING[16144]: config.c:1796 process_text_line: parse error: No category context for line 631 of /etc/asterisk/pjsip.conf
after fix global issue
2012 Sep 20
6
accept email and make phone call?
Any ideas on how asterisk could accept an email (such as an email to SMS or "number at mybox.org" sort of thing) and make a phone
call to a specific number and make an announcement?
I imagine the first part is the big question.
joe a.
2009 Dec 29
1
identifying channel for softhangup
When I place an outbound call from asterisk 1.6.1.12 to a FXO port on
my Cisco 1760V 12.4, the channel changes - seemingly incrementing:
e.g., in the first call, below, the channel name is
"SIP/vgw1-00000075" -- the second call (on the same FXO port after a
soft hangup on the CLI) is "SIP/vgw1-00000077"
How can I extract this information in the dialplan so that I can use
2011 May 13
1
asterisk 1.8 + google voice
somewhere along the way, i noticed incoming calls from google voice are
no longer working on my asterisk 1.8.3.2 system.
When the call comes in, asterisk immediately prints on the console:
== Spawn extension (google-in, s, 2) exited non-zero on
'Gtalk/+12153930924-f947'
[May 12 22:47:18] NOTICE[13835]: chan_gtalk.c:1977 gtalk_parser: Remote
peer reported an error, trying to