Displaying 20 results from an estimated 5000 matches similar to: "calls ending up in default context"
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
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
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
2011 Feb 04
1
SoftHangup on asterisk 1.8.2.3
I am trying to use SoftHangup in my dialplan, but it's either not
working or I'm not using it correctly.
when i'm on the console, i see:
pbx1*CLI> core show channels
Channel Location State Application(Data)
SIP/vgw1-000000a2 2156181505 at inbound:1 Up AppDial((Outgoing Line))
SIP/143-0000009f s at macro-SaferSIPDial Up Dial(SIP/99302156181505 at vgw1,,
2 active
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
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:
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 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./
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
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
2009 Nov 18
1
clever ways to "share" an extension between sip and fxs
Using Asterisk 1.6.1.9, I'm looking for a way to "share an extension"
between a SIP phone (Cisco 7940) and a SLT on a FXS port of a Cisco 1760
(via sip) -- at any given time I want to be able to pick up either phone
and it should be "bridged" to the other - just like having two SLTs on the
same copper pair.
The goal is to have a cheap cordless telephone sit right next
2003 Sep 26
1
Cisco 2600 and ASTERISK and calling out
You have no dial-peer telling the router what to do with the outbound call.
http://www.tape.net/~gerry/asterisk/cisco26x0.html
At 12:50 PM 9/26/2003, you wrote:
>Like Gerry wrote for callerid you need VIC-2FXO-M1 card.
>
>Right now I am stuck on making outgoing call.
>
>Could soembody help me with the configuration.
>
>On cisco I have soemthing like that:
>
>dial-peer
2006 Jun 12
0
tftp trouble
I am trying to tftp from a cisco router on my lan, but it is not
working. All I see are these errors in the log. I can "tftp
localhost" without a problem as well. Any suggestions on what I can
check next? I don't see too much troubleshooting available search via
google.
Thanks.
[root at 170 chadr]# uname -a
Linux 2.6.16-1.2111_FC4 #1 Sat May 20 19:59:40 EDT 2006 i686 i686
i386
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: ${SWIFT_DTMF})
if while I am listening to the playback, i interrupt and dial:
- "12345", SWIFT_DTMF is set to
2003 Mar 02
0
Malformed/Missing Contact field??
Anyone know what this message means?
-- SIP/2113-945d answered SIP/111.222.96.14:5060
-- Attempting native bridge of SIP/111.222.96.14:5060 and SIP/2113-945d
-- Got SIP response 400 "Bad Request - 'Malformed/Missing Contact field'" back from 111.222.96.14
111.222.96.14 is a Cisco 1750 with a VIC-2FXS and a VIC-2FXO.
The call comes into the Cisco, gets sent to the
2013 Feb 12
1
asterisk 11 AGI
I recently upgraded to asterisk 11 from 1.8.
I had VXML working via AGI in 1.8 - from extensions.conf:
[VXML]
exten => s,1,Answer
exten => s,n,Set(ENCODED=${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]
2015 Apr 29
0
PJSIP - sessions-timers support not working on 13.X
Ok , digging more into this i could see that (timers=no) and (timers=forced) not work asterisk not pay attention to this options when is reloaded cli not say anything and when the pjsip show endpoint <endpoint> it show always timers=yes when (timers=no) and (timers=forced) to that endpoint.
I wonder to force asterisk to refresh the session in some cases when is needed .
pjsip is able to
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
2007 Sep 13
4
[CentOS 5] tftp-server, unable to create new files (even with "-c" option)
Hi all,
I'm trying to setup a TFTP server to serve as repository for the
config of all my Cisco network devices.
As per the the tftpd man, I've added the "-c" option into the /etc/
xinetd.d/tftp (as follows) but I still cannot get write access
(unless the file is already present).
[root at chl1 ~]# cd /etc/xinetd.d
[root at chl1 xinetd.d]# cat tftp
# default: off
#