Displaying 20 results from an estimated 4000 matches similar to: "One Way Audio from External Sip Soft & Hard Phone"
2011 Feb 18
2
no progress indication
I am integrating a new server (Asterisk 1.8.2.3, DAHDI 2.4.0) with VOIP
only trunks, and this server only has soft phones.
When I dial an extension and the phone is not registered, I don't get any
ring or progress indications, and eventually the Dial() times out and
drops into voicemail (as expected).
CLI output:
-- Executing [s at macro-StdExten:6] Dial("IAX2/barneveld-2036",
2011 Apr 04
2
WARNING chan_sip.c:3115 __sip_xmit
Hey Guys,
Whenever i calling any extension i am getting following WARNING messages do you have any idea they coming from where?
-Satish
shirley*CLI>
== Using SIP RTP CoS mark 5
-- Executing [7623 at from-sip:1] Macro("SIP/7527-00000008", "stdexten,7623,sip/7623&sip/7624") in new stack
-- Executing [s at macro-stdexten:1] Dial("SIP/7527-00000008",
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
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 =>
2007 Jun 12
2
Bridge bug in 1.4?
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
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",
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?
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:
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} -
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
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,
2003 Dec 13
2
Wrong voicemail after transfer?
I'm using a modified "default config" file for extensions.conf, the one
that uses macro-stdexten to handle the stations.
We use a TDM30 card for our stations.
When a call that has been rung in using that macro transfers the call
things work just fine as far as the "other" instrument ringing.
But once the ring timeout has expired, the call then drops into the
*original
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
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 =>
2011 Apr 21
3
missed call notification
Hi,
I am looking at http://www.theschmandts.org/blog/?p=28 to setup missed call notification but i am having issue. following is my dialplan
[macro-stdexten]
exten => s,1,Dial(${ARG2})
exten => s,2,Goto(s-${DIALSTATUS},1) ; Jump based on status (NOANSWER,BUSY,CHANUNAVAIL,CONGESTION,ANSWER)
exten => s-NOANSWER,1,Voicemail(${ARG1},u) ; If
2004 Dec 10
2
include and hint in extensions.conf with new realtime feature - how?
hi,
i'm a bit puzzled because i do not get include and hint to work with the
new realtime enginge (cvs-head from 2004-12-09).
other things (sipfriends and "normal" extensions) work perfect with the
realtime engine.
the entries in the static extensions.conf file i used before where:
exten => 183,hint,SIP/snom220
exten => 183,1,Macro(stdexten,443,SIP/snom220,183)
exten =>
2009 Jul 20
1
callforward with asterisk-gui.problem with stdexten
Hello, i am trying to enable call forwarding on asterisk 1.6 with
asterisk-gui
If i set my stdexten as follows (with the lines i marked) everything seems
like working.
But if i make any change on asterisk-gui and apply it.. it recreates the
macro-stdexten and deletes my configuration regarding to it.
So where should i add my call-forward configuration???
Where am i making a mistake??