Displaying 20 results from an estimated 1000 matches similar to: "one extension goes straight to voicemail, others don't"
2003 Apr 25
1
Fwd: [bug & patch] libstdc++ in linux_base-7.1_3
Hey guys, hope you don't mind this forward email, because I have been
reading the 'flashpluginwrapper problem.' subject. The bug is in the
linux_base, but nobody seem to check on this in the freebsd-ports. Looks
like Brandon D. Valentine might needs to create a PR to get the better
attention and keep the track.
Cheers,
Mezz
>Summary:
>
>This began as a problem report
2005 May 13
0
Extension never ring, goes straight to VM
I installed a new * server and copied the sip.conf and extensions.conf
from my existing setup to the new box. I created my outbound trunk
with a different broadvoice account and am able to dial out without
issue. I am able to dial all extensions but I go straight to voicemail
without any ringing, this happens to both local and outside numbers
calling my extensions. I have the correct firewall
2004 Jan 07
1
Call Rollover
Have a question about implementing Call Rollover with my current
extensions.conf configuration.
[macro-stdexten]
exten => s,1,Dial(${ARG2},20) ; Ring the interface, 20 seconds
maximum
exten => s,2,Voicemail2(u${ARG1}) ; If unavailable, send to voicemail
w/ unavail announce
exten => s,3,Goto(default,s,1) ; If they press #, return to start
exten =>
2012 Oct 31
1
Asterisk 11 and stdexten written in AEL invoked by pbx_config
Almost two years ago, a change between how AEL code is built into
Asterisk dialplan between minor versions made clear the need to
provide a sane entry point into AEL subroutines and that's how
AELSub() born.
With Asterisk 11 release, they way [stdexten] at extensions.conf is
invoked changed from Macro to Gosub using the 'missing context
feature' and this caused that any stdexten
2008 Mar 13
3
Newbie One-touch Recording: Does not work (more info)
I thought it was quite easy to implement but I cannot get one-touch
recording to work. Here are the changes what I did:
I restarted Asterisk after the change (because reload does not work for
changes in features.conf).
I press *1 on the Polycom IP600 phone to record a conversation but no
new wav file appear in /var/spool/asterisk/monitor or elsewhere.
Test A: Outside line calling in
2011 Apr 03
1
From 1.4 to 1.8: stdexten issue
Hello all,
I'm in the middle of upgrading my asterisk setup to 1.8 (1.8.2.3) and
I'm completely confused by the gosub/stdexten thing.
I used to call the stdexten macro but I haven't been able to figure out
how to use Gosub.
I'm using the sample extensions.conf and added something like this:
=========================
[home]
include => stdexten
exten =>
2006 Oct 30
0
Realtime trouble with contex
Hello, Asterisk.
I am currently using Asterisk (asterisk-1.2.13) and asterisk-addons-1.2.3_1
on FreeBSD 6.1-RELEASE-p10
So, after setup asterisk for realtime extension:
res_mysql.conf:
[general]
dbhost = 127.0.0.1
dbname = asterisk
dbuser = asterisk
dbpass = asterisk
dbport = 3306
dbsock = /tmp/mysql.sock
res_odbc.conf:
[mysql]
enabled => yes
dsn => asterisk
username => asterisk
2006 Apr 05
0
What does this error mean "app.c: Huh....? no dial for indications?"
Hi,
What does the following error mean:
Apr 5 12:39:40 NOTICE[22755] app.c: Huh....? no dial for indications?
Here is the 'full' log around the error:
Apr 5 12:38:24 VERBOSE[22755] logger.c: -- outgoing agentcall, to
agent '3002', on 'Local/510@default-6b6c,1'
Apr 5 12:38:24 VERBOSE[22755] logger.c: -- Called Agent/3002
Apr 5 12:38:24 VERBOSE[22755]
2005 Feb 01
0
Troubles with Macro-stdexten and dial
Hi!
Could someone give me a hand?
If I dial 200 for echo testing it works... Everytime I dial an extension ex.
505 get the error below....
In this example it was from 508>505 a Xlite Pro to a TA.
I believe it has something to do with the way i'm executing the command dial
but I use the "standart" that comes in the samples from asterisk.
*CLI> -- Executing
2006 Oct 25
1
Phone Rings, Immediate Hangup and then Rings Again.
I am having a problem with an Asterisk server, in that when it is
receiving a call from another Asterisk server using an IAX2 trunk the
phone rings for 10 ms and then there is a hungup from asterisk and then
the phone rings again before another hangup.
The funny thing is that after I really hang up on the calling phone it
repeats this as if I am still trying to call.
Any Ideas?
2006 Jan 26
4
extension to extension dialing
Sorry for all the newbie questions. I really appreciate everyone's help
today.
Okay I've got outgoing and incoming calls working with no echo. yay! Now
I'm having an issue with SIP extension to extension calling. Any time I
dial another extension it goes right into voice mail. My
extensions.conf is pretty small and rough but, here's what I have right
now. Most of it was taken
2009 Mar 03
2
macro-stdexten question
I am running asterisk 1.4 and the Digium GUI SVN-branch-2.0-r4489.
When one phone calls another, I see the following on the console
(here, 6223 dials 6123)
-- Executing [6123 at DLPN_DefaultDialPlan:1] Macro("IAX2/6223-10489",
"stdexten|6123|SIP/6123&IAX2/6123") in new stack
-- Executing [s at macro-stdexten:1] Set("IAX2/6223-10489",
2014 Oct 23
1
logger.conf
with the below defined in logger.conf on 11.6 cert 6
I am not getting any log message other than notice and warning in any files
when doing module reload logger - queue log is the only one that says it
restarts
*CLI> module reload logger
== Parsing '/etc/asterisk/logger.conf': Found
Asterisk Queue Logger restarted
built fresh box with make samples - added 2 stations, dialing from
2003 Jul 14
0
Cisco 7960 Transfer Call drop problem
Hi,
I'm having problems with transfer from an analog line via a X100p and Cisco
7960's running SIP.
With an attended transfer the a call comes in, I transfer it to another
7960, they answer I announce the call, press transfer again, the two parties
talk for 1-2 seconds then the analog line drops, though the Cisco phone is
not aware of this, i.e. nothing on the screen changes. The
2015 Mar 26
1
CDR dst value null after attended transfer
I'm having an issue with CDR. Basically, I expect to have all "legs" of a
call having the same linkedid and differing only by the sequence value.
That does happen, but I'm getting null dst values after doing an attended
transfer.
I'm not sure if this is a bug or I'm doing something wrong. I'm running
Asterisk 13.2.0.
Here's the console log, step by step:
First,
2006 Apr 05
2
What causes deadlock?
Hi
What causes deadlock?
Apr 5 14:02:43 WARNING[2413] channel.c: Avoided initial deadlock for
'0x82acb10', 10 retries!
Apr 5 14:02:43 WARNING[2413] channel.c: Avoided initial deadlock for
'0x8298160', 10 retries!
Here is the portion of the log:
Apr 5 14:02:42 NOTICE[23363] chan_zap.c: Got event 18 (Ring Begin)...
Apr 5 14:02:42 VERBOSE[23363] logger.c: -- Executing
2010 Apr 29
2
Code in extensions.conf to leave a voice mail in another PBX ?!
Hi Guys,
i spent some time to figure this out (since i love how dialplan is written)
but i decided to ask for your help guys.
i have two asterisk servers one is 1.2 the other is 1.4, from 1.4 (pbx1) to
1.2 (pbx2) i can leave a voice mail without any pb, but from pbx2 to pbx1 it
just hang up.
in pbx2 extensions.conf:
i am using: exten => 8029,1,Dial(IAX2/pbx1/${EXTEN},20,tTWwr)
in pbx1, i
2008 Mar 13
5
Newbie One-touch Recording: Does not work
I thought it was quite easy to implement but I cannot get one-touch
recording to work. Here are the changes what I did:
I restarted Asterisk after the change (because reload does not work for
changes in features.conf).
I press *1 on the Polycom IP600 phone to record a conversation but no
new wav file appear in /var/spool/asterisk/monitor or elsewhere.
Any suggestions?
Here is the console log:
2006 Mar 19
0
Transfer to specific park number
Hi
I'd like to allow users to transfer a call to a specific park number. This
way, the receptionist can tranfer a call park for ext 100 at park number
7100 etc...
It seems like this should be fairly simple using the Park(ext) app but it
doesn't work for me. No matter what I extension I use, the system just
picks the next available park number. I've simplified my dialplan for
2003 Dec 30
2
playback in [macro-stdexten] problem
I added the playback line to my [macro-stdexten] context but when I dail
an extension I don't get the "please hold while I try that extension"
message. It just dials the extexsion. Do I have a syntax problem
somewhere ?
exten => 8005,1,Macro(stdexten,8005,Zap/2)
exten => 8006,1,Macro(stdexten,8006,Sip/8006)
[macro-stdexten]
;
; Standard extension macro:
; ${ARG1} -