SoftHangup on asterisk 1.8.2.3

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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@inbound:1 Up AppDial((Outgoing Line))
SIP/143-0000009f s@macro-SaferSIPDial Up Dial(SIP/99302156181505@vgw1,,
2 active channels
1 active call
194 calls processed
pbx1*CLI>
in my dialplan, i have:
exten => s,1,Set(CHAN=${SHELL(asterisk -rx "core show channels" | awk
'/^SIP/vgw1-/ { print $1 }' | head -1)})
exten => s,n,SoftHangup(${CHAN})
exten => s,n,Wait(2) When I dial…

Asterisk Users 4.5 years ago 2 Answers

Queues and Agent penalty - how to go to second best agent when the first does not answer

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>
> I am a little confused as to what the OP wants the system to do? Call the
> proper agent, but when they don't answer, on the next call, it shouldn't
> call the same agent? OK, but for how long? 5 minutes? Until they manually
> unpause (current option as described by Kevin), 30 minutes? Should it then
> up their penalty? For how long? I should have been more precise. I don't actually expect all this to
happen, but here's what I wish it did: 1) Ring agents in Round…

Asterisk Users 4.5 years ago 0 Answers

MP3 Crashing Asterisk

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Hi Users, I have a problem with some of my mp3 files. they crash the system
(Asterisk 1.6.2.14 on a x86_64 running Fedora 13 ) when it tries to
play them. Unfortunately the logs do not give me a clear fault or
cause of crash but i can clearly see that ts because of the MP3 files.
Its the way some files are encoded. Is there a way I can make it skip
the files that can be played? I use the Playback() and Background()
Applications (Not MP3Player) Has anyone experienced this before? I…

Asterisk Users 4.5 years ago 1 Answer

problems with voicemail and centos 5

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i have installed asterisk 1.8 following this doc
http://www.asterisk.org/downloads/yum i installed the package
asterisk18-voicemail-imapstorage-1.8.2.2-1_centos5
in order to store voicemail in imap but the application voicemail is not available when i type
core show application ? in the asterisk log file i have these messages
[Feb 3 19:00:20] WARNING[14311] loader.c: Error loading module
'app_voicemail_imapstorage.so': /usr/lib/libc-client.so.1: undefined
symbol: mm_dlog
[Feb 3 19:00:20] WARNING[14311] loader.c: Module
'app_voicemail_imapstorage.so' could not be loaded. does someone know how to solve that problem?
grocanar
Newsterisk Posts: 1
Joined: Thu Feb 03, 2011 3:44…

Asterisk Users 4.5 years ago 0 Answers

voice quality measurement using dahdi_monitor

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hi group , i am working on dahdi_monitor for measuring voice quality , so i want to
know that on which data i can tell that this PRI
lines are working properly, is there any measurement on basis of that i can
make MOS. i am working from last 2-3 days
but i only get idea about making .raw file and making .wav file and visulal
mode of RX and TX of PRI line. what i want is measurement of voice quality so that i can talk with provider
that i am getting %…

Asterisk Users 4.5 years ago 2 Answers

Outgoing FXO calls have no audio with callprogress=no

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> My outgoing FXO calls are answered but have no audio in
> either direction if I have callprogress=no in
> chan_dahdi.conf. If I change to callprogress=yes then the
> audio returns. My chan_dahdi.conf file is listed below. Can
> anyone point-out why callprogress=no isn't working?
>
I'm assuming your telco doesn't support line reversal on answer, you need to
set answeronpolarityswitch=no Hope that helps

Asterisk Users 4.5 years ago 0 Answers

Email alerts for trunks (peers)

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Hey Guys, I'm after a way to monitor our sip trunks (peers) and send an email if they go down. I know I could use 'asterisk -rx "sip show peers"' in a shell script but that seems messy, especially since I'd like to monitor it fairly closely (so I'd like to run it every 20 or 30 seconds or so). Is there a better way to do it?

Asterisk Users 4.5 years ago 2 Answers

PRI voice optimization

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Hi All, This posting regarding PRI voice optimization, on dahdi 2.1.0.4. we have more than 4 machine running on 4 port PRI card with echo
cancellation hardware based. i have enabled echo cancel from chan_dahdi.conf using echocancel=yes, now
more than 70% of call get good voice
but some of calls having issue for callquality and other voice related
issues. now my question is that is there
any voice related parameter that we need to set for INDIA specific region
and is ther any voice hardware tester for PRI
that we can use and…

Asterisk Users 4.5 years ago 5 Answers

Outgoing FXO calls have no audio with callprogress=no

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My outgoing FXO calls are answered but have no audio in either direction
if I have callprogress=no in chan_dahdi.conf. If I change to
callprogress=yes then the audio returns. My chan_dahdi.conf file is
listed below. Can anyone point-out why callprogress=no isn't working? #cat /tmp/a
[trunkgroups] [channels]
language=en
context=incoming
toneduration=40
;usedistinctiveringdetection=yes
answeronpolarityswitch=yes
usecallerid=yes
cidsignalling=bell
cidstart=ring
;hidecallerid=yes
;hidecalleridname=yes
;waitfordialtone=yes
;mwimonitor=no
;mwilevel=512
;mwimonitornotify=/usr/local/bin/dahdinotify.sh
;mwisendtype=rpas,lrev
callwaiting=yes
;restrictcid=no
usecallingpres=yes
sendcalleridafter = 1
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
canpark=yes
cancallforward=yes

Asterisk Users 4.5 years ago 0 Answers