Asterisk as a caller ID

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In a 1.4.24 system, out of several lines, one of ours gets 1 or more random
calls a day with "Asterisk" as the caller ID. I have just seen this described in the last couple of weeks, but at the time
it wasn't happening to us, and I the explanation didn't stick with me. Can anyone give me a pointer to this "feature"? Searching the message base
for "Asterisk" seems futile. Thanks! Cary Fitch

Asterisk Users 4.7 years ago 1 Answer

Friend/user/peer in plain English?

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Hello I've done some googling, but still puzzled at my working
configuration. Apparently, a "user" can only receive calls through Asterisk, a "peer"
can only make calls, and a "friend" can do both. If that's correct, I don't understand why my VOSP requires the
following settings in sip.conf to let my Asterisk server make/receive
calls to/from the PSTN: =============
[general]
...
register => me:pass@vosp.com [vosp_outgoing]
type=peer
host=vosp.com
username=me
secret=pass
fromuser=me
fromdomain=vosp.com
nat=yes
canreinvite=no
qualify=yes [vosp_incoming]
;why not type=user?
type=peer
host=vosp.com

Asterisk Users 4.7 years ago 0 Answers

hint not working

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I'm trying to migrate from MeetMe to ConfBridge: [conferences]
exten=>_8[1-9],1,Answer()
;;exten=>_8[1-9],1,Meetme(${EXTEN},1Ms,1234)
exten=>_8[1-9],2,ConfBridge(${EXTEN},1Ms)
exten=>_8[1-9],n,Hangup
And that works. Also changed the hints: ;;exten => 81,hint,MeetMe:81
exten => 81,hint,ConfBridge:81
;;exten => 82,hint,MeetMe:82
exten => 82,hint,ConfBridge:82
;;exten => 83,hint,MeetMe:83
exten => 83,hint,ConfBridge:83
;;exten => 84,hint,MeetMe:84
exten => 84,hint,ConfBridge:84 And that does not work. The blf does not go on when a party is in
ConfBridge. Is there some new syntax for hints with ConfBridge? sean

Asterisk Users 4.7 years ago 4 Answers

Asterisk 1.6 produces *many* zombie processes on Debian.

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Am 20.12.2010 21:39, schrieb Ernie Dunbar:
> We have an issue with our Asterisk install where Asterisk produces many
> Zombie processes (on the order of several hundred per minute) until either
> the Asterisk server is restarted (and the zombies die a natural death), or
> the kernel runs out of PID space (happens within hours) and brings the
> system to a halt.
>
> This problem only happens when the server is under some non-trivial load.
> We were testing this server with 8 SCCP phones, making up to…

Asterisk Users 4.7 years ago 3 Answers

Setting `userfield` from within a callfile

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On Monday 20 Dec 2010, Olivier wrote:
> 2010/12/20 A J Stiles
>
> > Our current Asterisk 1.6.2.9 setup includes a CGI auto-dial application
> > (written by someone else before me) which sets up calls by creating
> > files of
> > the general form
> >
> > Channel: SIP/$INSIDE_NUMBER
> > Context: $CONTEXT
> > Extension: $OUTSIDE_NUMBER
> > Priority: 1
> > CallerId: $INSIDE_NUMBER
> >
> > in /var/spool/asterisk/outgoing/ .
> >
> > It works…

Asterisk Users 4.7 years ago 0 Answers

app_voicemail.c how to enable debugging?

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Hi Looking at the source of app_voicemail.c there are many statements like: ast_debug(1, "%s doesn't exist, doing what we cann",
prefile); Where do I have to enably this to be showed in the console or logged to a file
by logger. core set debug does not seem to help here. Well, my actual problem is, that if a customer has recorded his own greeting,
he usualy tells the caller to record his message after the tone, so
app_voicemail should not play the intro. spool/mailbox/unavail.gsm
vm-intro.gsm
beep.gsm but only spool/mailbox/unavail.gsm
beep.gsm In case there…

Asterisk Users 4.7 years ago 1 Answer