When I mv a file to /var/spool/asterisk/outgoing in order to place a
call from a user extension that will always be recorded, what
parameter do I set in the call file in order to specify an exact name
for the wav file?
This is on Trixbox and at the moment I'm considering setting an extra
variable
We're having some difficulty tying together our Cisco and Audiocodes
syslogs with our Trixbox asterisk logs.
We'd like to have some way to split out a single call from all the
activity going on at one moment.
Obviously NTP is the first step for this, but we haven't found any
means to tie the
On Tue, Apr 1, 2008 at 11:03 AM, Mike Trest - Personal [EMAIL PROTECTED]
wrote:
At 01:13 PM 4/1/2008, you wrote:
Is app_conference stable now?
I've never made it through a thousand calls without a crash. (With a
busy call center this doesn't take all that long.)
I have deployed a
On Tue, Apr 1, 2008 at 7:29 AM, Chaya Zipora Rosenberg
[EMAIL PROTECTED] wrote:
Hello,
I am setting-up a system to place outgoing calls for a certain
number of minutes (as allowed per the customer's account). I would
like to break into the long distance channel to announce 1 minute
On Wed, Mar 12, 2008 at 1:57 PM, Michiel van Baak [EMAIL PROTECTED] wrote:
On 16:27, Wed 12 Mar 08, Steve Totaro wrote:
Try Callweaver.
Thanks,
Steve Totaro
or app_conference for asterisk.
That does the trick for me on OpenBSD where you dont have
ztdummy.
Is app_conference
Out of stock now.
Any war stories about running Asterisk on a serious blade setup?
Will you ever hire Wesley Snipes to flog them at a convention?
-HJC
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asterisk-users mailing
Local (Indian) vendor, Intel(R) Pentium(R) 4 CPU 2.66GHz, 1 GB RAM,
Two 80GB IDE disks, X100P clone.
Supports 15 agents as SIP/ulaw to IAX/G.729 bridge running Vicidial
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On 7/16/07, Dovid B [EMAIL PROTECTED] wrote:
Define health. I was working on but gave up on it (no time) to have serverA
call serverB. ServerB has an agi that it runs that stores info in DB. if
serverB doesn't get a call then we know that there are issues (and run the
script vice versa).
[Jul 2 09:31:16] VERBOSE[2682] logger.c: == Refreshing DNS lookups.
[Jul 2 09:31:16] NOTICE[2682] dnsmgr.c: host 'outbound1.vitelity.net'
changed from 64.2.142.17 to 64.2.142.29
[Jul 2 09:31:23] DEBUG[2711] jitterbuf.c: Attempting to exceed
Jitterbuf max 600 timeslots
And the calls are
On 6/6/07, John Novack [EMAIL PROTECTED] wrote:
Henry Cobb wrote:
Why would anybody plug a telephone line into an X100P clone?
???
What else would one plug into it?
We just use them as clock cards for MeetMe and trunking.
-HJC
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--Bandwidth
On 6/7/07, Ricardo Martins [EMAIL PROTECTED] wrote:
We use G.729. Consumes only 35kbps of bandwidth and has a level 4 (from
0 to 5) of voice quality. We still have very poor public data networks
here in Brazil that makes G.711 a very high bandwith consunption codec
for us.
35kbps sounds very
On 6/6/07, carl Lougher [EMAIL PROTECTED] wrote:
Hi Chaps,
Is there a way to detect/highlight poor quality voice
calls going through an asterisk server?
Was thinking of picking up a cdr record or some other
variable showing poor quality on calls when the
internet is having issues.
Is there any
On 6/5/07, Jared Smith [EMAIL PROTECTED] wrote:
Most of the clone cards don't support far-end disconnect supervision,
so you'll have problems where Asterisk can't tell that the other party
has hung up the phone. You'd be better off to buy a modern Asterisk
telephony card.
Why would anybody
On 6/6/07, Matt [EMAIL PROTECTED] wrote:
I chart VNAKs per hour.
Would you care to share how you accomplish this? What programs do you use?
grep VNAK /var/log/asterisk/full | cut -d ' ' -f 4 | cut -d : -f 1 | uniq -c
Needs a bit of an adjustment between the 1-9th and 10th-31st of the
On 6/5/07, Ronaldo [EMAIL PROTECTED] wrote:
Hi all,
I'm planning to buy a X100P clone and would like some feedback about
this card.
Does anyone already used this card? Does anyone recommend it ? or not?
I'd appreciate any comments.
If you have a new 3.3v only motherboard then make very sure
On 6/5/07, Iban Lopetegi Zinkunegi [EMAIL PROTECTED] wrote:
Hi All!!
I have my asterisk working in my house (working with mandriva 2007 and
asterisk 1.4 svn). I´ve looking on the net but i couldn´t find the way of
making work my asterisk in a real enviroment. Seems that the problem of NAT
is a
Would anybody mind if the the following command options where added to
MeetMeAdmin?
0 - 9, * and #
I'm considering hacking the code to add these commands to play the
DTMFs to the specified user as tones and hope that the SIP or IAX
channels then work with these correctly.
-HJC
On 3/21/07, Louis-David Mitterrand
[EMAIL PROTECTED] wrote:
Hi,
At one location we have a user whose Polycom IP430 suffers from erratic
reboots. We swapped his phone for a brand new one, but the problem
remains.
Has anyone seen that?
Our Polycom 3s and 5s ship with flaky power supplies and
On 3/20/07, Olivier [EMAIL PROTECTED] wrote:
Let's say you have an Asterisk server running.
Which parameters would you check to improve service continuity ?
The tools I tend to use are vmstat, iftop (all VoIP, all the time),
show registry and df.
-HJC
On 3/15/07, Andrew Kohlsmith [EMAIL PROTECTED] wrote:
Obviously you didn't read Google's research paper on drive failures.
This one?
http://labs.google.com/papers/disk_failures.html
-HJC
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On 3/15/07, Chris Nighswonger [EMAIL PROTECTED] wrote:
Ok. I have not been able to setup the box to call outside, however,
watching the packet traffic I see plenty of data flowing from the
xlite client to the * server, but never any packets from the server to
the client. (That is, during the
On 3/12/07, Dave Cotton [EMAIL PROTECTED] wrote:
On Mon, 2007-03-12 at 20:52 +1100, Paul Hales wrote:
More importantly, how many calls per day and how long per call.
Then you can figure out the other bits.
He wants to make 50 simultaneous calls. What difference does the length
and frequency
On 3/13/07, Chris Bagnall [EMAIL PROTECTED] wrote:
His vindictive dialer isn't playing while it is listening to rings or
busy signals.
Forgive my ignorance, but what on earth's a vindictive dialer? Is it one
with a strong sense of revenge? :-)
A normal predictive dialer determines from
On 3/10/07, Lacy Moore - Aspendora [EMAIL PROTECTED] wrote:
On 3/10/07, Henry Cobb [EMAIL PROTECTED] wrote:
So get a second broadband connection and run only voice on it.
Has anyone tried this?
I have been thinking about this. We're getting so much spam that I
think it's taking up too much
It looks like somebody has fixed the X100P v3.3 problem in an outbreak
of sanity. (Pity that Digium seems to be in such a hurry to not take
my money. Well, other than the G.729 tax.)
On 3/9/07, Zeeshan Zakaria [EMAIL PROTECTED] wrote:
I am thinking of going with HWEC and also using a good QoS switch. Right now
there is only one switch (don't remember the name) and it is handling all
the VoIP and data traffic. Sometimes voice breaks, and it must be because of
interference
On 3/9/07, mail-lists [EMAIL PROTECTED] wrote:
[test]
disallow=all
allow=gsm ;GSM consumes far less bandwidth than ulaw
;allow=ulaw
;allow=alaw
Are you sure that the xlite phone can handle gsm??
I use it on Linux and it does.
-HJC
___
I've just moved into 3.3v PCI servers and found that my clone X100P
cards were lying about the 3.3v supported notch.
Can I use a Wildcard TDM400P without any modules as a timer for
MeetMe in a 64 bit 3.3v server?
Will I still need to plug the hard disk power cable into it?
Is there a better
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