Displaying 20 results from an estimated 1000 matches similar to: "useless mpg123 processes hanging around"
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
2004 May 24
5
mpg123
When I start * I get 6 mpg123 processes start as well. Is this normal?
Often after a couple of days these mpg123 processes start to consume cpu and
I have to kill them off.
I do not have a sound card in the server and I have noload => chan_oss.so
Simon
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 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
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 Aug 17
0
Trouble with call pickup using RPID with Cisco
I have a Cisco 1760V with FXO ports hooked to POTS lines talking SIP to
asterisk 1.8.15.0.
imagining in extensions.conf:
exten => 1,1,Dial(SIP/121)
exten => 2,1,Dial(SIP/121&SIP/122)
When a caller dials extension 2 /and/ I have
trustrpid=yes
generaterpid=yes
sendrpid=yes
in sip.conf and I use the pickup exten, the caller is disconnected.
see:
2009 Dec 18
0
calls ending up in default context
I'm trying to figure out how calls are ending up in my default
context (which should never happen).
I've got a Cisco 1760V with a VIC-2FXO-M1/VIC-4FXS and 5 Cisco sip
phones.
When I make a call from one of the FXS ports on the 1760, the call
goes into asterisk's default context instead of where i think i'm
directing it.
Can someone tell me what I have misconfigured?
1760
2004 Jan 10
1
default music source for SIP channel
The wiki says this about the MusicOnHold command:
"Plays hold music specified by class. If omitted, the default music
source for the channel will be used."
http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20MusicOnHold
How do I set the default music on hold class for the SIP channel ? I
tried adding musiconhold=test to my sip.conf.
musiconhold.conf looks like this:
2003 Oct 01
2
newbie question: MOH problem
Just the sort of newbie question we all hate ;-)
I'm a bit stuck with MOH. I think all is done right and I've read
everyhing I can find, but whenever * tries to do MOH, all that happens
is
'-z: No such file or directory'
Yes, I am on redHat. Yes I have installed real mpg123. Yes, it does
seem to work from the command line.
Any suggestions would be greta, I'm sire
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./
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]
2013 Jun 16
2
MOH don't work after update
Hi
we have a small problems.
We have a Asterisk 1.6.1 old server with music on old.
we have updated to AsteriskNow 11.4.0
and now, when we want play sound, we have a errors:
-- Executing [334xx at Accueil_HNO:2] BackGround("SIP/SIP000005-0000000c",
"Fermeture") in new stack
[Jun 16 07:35:06] WARNING[7634][C-00000006]: file.c:701
ast_openstream_full: File Fermeture does
2003 May 13
5
Music on hold, Call Parking, etc
Ok, this falls under the newbie category:
Has anybody created any documentation on using musiconhold or call parking?
I have found sample config files for musiconhold, but I'm not sure how they
work.
[musiconhold.conf]
[classes]
loud=>mp3:/var/lib/asteriks/mohmp3
How do I then set up this feature in extensions.conf?
I can't seem to find an example of what I'm looking for (or I
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
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
2004 Apr 08
1
Live Music on Hold
I have a small * system in my home (1 U100S, 1 X100P, 1 BT101, and 1 SPA2000) to handle my requirements. I would like to add Music on Hold and have been watching the forum to see if something would come across on this topic. The difference I am interested in is getting the music from a radio or someother external source. All references to MOH
up to now have been using MP3 files and going
2011 Jun 06
2
issues.asterisk.org
i'm trying to review issues that i'm monitoring and/or have reported at
http://issues.asterisk.org
when I click on 'My View' or 'View Issues' I get an error:
APPLICATION ERROR #401
Database query failed. Error received from database was #1142: DELETE
command denied to user 'mantisreadonly'@'localhost' for table
'mantis_tokens_table' for the
2005 Jun 29
4
Music oh hold
Sorry, i also tried this:
exten => 6000,1,Answer
exten => 6000,2,MusicOnHold(default)
and i got this result:
*CLI> -- Executing Answer("SIP/2391-8cdd", "") in new stack
-- Executing MusicOnHold("SIP/2391-8cdd", "default") in new stack
Jun 29 19:33:47 WARNING[1616]: res_musiconhold.c:354 moh0_exec: Unable to start music on hold (class
2011 Mar 05
0
[announce] jkSMS
For those interested, I have released a first version of jkSMS, which is
a simple package that lets cell phones text messages to "asterisk".
Note it's not real SMS, it makes heavy use of email-to-sms gateways, but
it seems to work well. I have had the code running > 12 hours, but
haven't found any issues.
it's not for the faint-of-heart and might require a bit of
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