How about avoiding the phone entirely in the playback phase?
Have asterisk record the call to disk in MP3 or Slin, then use a pc with decent
audio card
to read it off the shared disk and feed it to the mixer.
Tim.
- Original Message -
From: Bob Pierce [EMAIL PROTECTED]
To: Asterisk
You can also cat /proc/zaptel/1 and parse the output of that.
This doesn't tell you if asterisk is happy though.
For that you could use the asterisk SNMP (sub)agent
Tim.
- Original Message -
From: Jerry Geis [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: Sunday, April 13,
Do your callees have known IP addresses ?
If not, how does Asterisk know where to send the call?
Normally this works because the callee has registered with asterisk,
so asterisk has an IP address to associate with the name (or number).
Without this asterisk can't know where to send the packets!
try setting
transfer=no
or
notransfer=yes
in iax.conf
Depending on the age of your asterisk version.
Tim.
- Original Message -
From: randulo [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: 20 February 2008 18:12:01
Isn't that what the IAXy uses to load new firmware ?
If so, asterisk probably supports it. (and the IAXy of course).
Tim.
- Original Message -
From: Cavalera Claudio Luigi [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent:
Jared was talking about a decent IAX hardphone on this list a week or so back,
I don't recall the make.
If you use IAX, all you need to do is :
1) set your local firewall to forward udp 4569 to asterisk.
(optionally filtering by from IP address if your user has a
fixed IP address or known
There is an option you might consider (if you are starting from scratch).
Don't use citrix. Write a web app.
Then embed a softphone in that web app.
Tim.
___
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
asterisk-users
Please read:
http://www.voip-info.org/wiki/view/FreeSwitch
and
http://www.voip-info.org/wiki/index.php?page=Asterisk
Then if you have a specific question about one of them,
come back here to ask about asterisk, and on the freeswitch
mailinglist for more info on that technology.
Or you could
- Original Message -
From: Vincent [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com
Sent: 16 January 2008 22:01:55 o'clock (GMT) Europe/London
Subject: Re: [asterisk-users] [IAX] Up-to-date list of soft- and hardphones?
On Wed, 16 Jan 2008 18:08:23 + (GMT), Gordon Henderson
You could hack it up by dropping them both into the same conference.
You'd have to tweak the messages and other conference settings, but it would
certainly work. Not as efficient as bridging though.
Tim.
- Original Message -
From: Douglas Garstang [EMAIL PROTECTED]
To:
If you search for Asterisk res_snmp on suse 9.1 in the page
http://www.voip-info.org/wiki/view/Asterisk+monitoring
You will find my notes. (Weirdly formatted I'm afraid)
I downloaded net-snmp-5.3.0.1.tar.gz and built/installed that.
Tim.
- Original Message -
From: yonoko molomo [EMAIL
Yes, res_snmp seems to be sensitive to the specific version of net_snmp.
I wrote some notes on this - see
http://www.voip-info.org/wiki/view/Asterisk+monitoring
Basically I ended up installing netsnmp from source, and things started working.
I'm currently writing a little demo program which
You can also monitor active calls with SNMP. This should be a lower load
than manager (I've no stats to prove it). It is also easier to secure as
the asterisk MIB is read only :-)
I'm in the process of writing a proof of concept that monitors to Asterisk
over snmp.
Tim
- Original Message
I guess that you just need to add a rule to your simple router's config
that permits udp 4569 from asterisk outbound to any IP address.
Tim
- Original Message -
From: Perssy Llamosas [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
I'm not sure what you want to do.
Registration is normally done by phones,
(or other pbx's) - asterisk java is more of
a call control suite.
You can use it to route calls to endpoints,
but you need to have the endpoint registered
with the PBX first (or on a known IP address)
Tim.
- Original
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