* You are viewing the archive for February 4th, 2011

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@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 the extension to invoke the above dialplan code, the console
shows:

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

>
> 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 Robin fashion, but always in the same order (could
simply use the already existing penalty value)
2) Always start from the top (taking into account the “ringinuse” value)

Basically, a simple _pre-ordered_ Roundrobin.

I could make this even better by (as you hinted at yourself) by using
autopause and asking for an “autounpause after x minutes” feature. But
those two things above would be wonderful, and I was actually surprised that
it wasn’t a possible setting. Unless I can order the agents somehow, but I
seem to understand that dynamic agents are sequenced in the order in which
they joined the queue, not according to some easily defined position value.

How I would envision this being configured? A queue setting that would
define how it handles penalty. Either in the current “Ring the best
agent(s) over and over again” or “try the good agents first, but then move
on”. Just a “yes/no” value.

Mike

MP3 Crashing Asterisk

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 searched the archives but the
few posts are all for way back in 2003, so they are not so helpful.

I also tried converting the files to wav or sln but there is severe
music quality loss. Anyone knows a relieable way of converting the
files?

Thank you!
Tim

problems with voicemail and centos 5

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 pm

voice quality measurement using dahdi_monitor

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 % of voice quality.i am sure there is
some better way to solve or debug .raw file and taking a decision.

let me help please to solve and finding problem of voice quality.

Outgoing FXO calls have no audio with callprogress=no

> 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