* You are viewing the archive for December 21st, 2010

Asterisk as a caller ID

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

Simplifying dial-plan

Is there a way to include:

_NXXNXXXXXX
_NXXXXXX
_011.
_911

into my current plan:

What is equivalent function to “mv” command in php for Asterisk Spool directory usage?

Hi Everyone,

I understand that there are a few warnings about using “cp” to move .call
files to /var/spool/asterisk/outgoing as they might acted upon before copy
is done. So, using PHP, What is the equivalent of “mv” command? Would it be
rename() in php or is there a better method?

Thanks,

Friend/user/peer in plain English?

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
context=from_vosp
nat=yes
canreinvite=no
insecure=port,invite
qualify=yes

[6011]
type=friend
secret=pass
context=my-phones
host=dynamic
qualify=yes
nat=no
=============

I would expect [vosp_outgoing] to be of type=peer, while
[vosp_incoming] should be type=user.

As a side-note, why do we need both a “register” and “fromuser/secret”
to make calls through a VOSP?

Thanks for any hint.

hint not working

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 1.6 produces *many* zombie processes on Debian.

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 five
> simultaneous calls through the DAHDI interface (a Digium Wildcard
> TE410P/TE405P (1st Gen)). Once our customers (nearly all SIP clients)
> start logging on and we get around 7 or 8 simultaneous DAHDI calls,
> Asterisk starts producing zombie processes at a high rate.
>
> We are using the following software:
>
> Debian Lenny 5.0
> Asterisk 1.6.2.15
> `dahdi show version`: DAHDI Version: 2.4.0 Echo Canceller: MG2
> Libpri 1.4.11.4
>
> A2Billing is also installed on this server, if that matters at all.
>
> Any help with this issue, including help in troubleshooting the cause, is
> highly appreciated.

What does /var/log/asterisk/messages say? And /var/log/syslog?