Javier Rios
2003-Nov-07 06:48 UTC
[Asterisk-Users] RE: msgs archives gsm of asterisk ??? Asterisk-Users digest, Vol 1 #1809 - 16 msgs
Hello.
The procedure so that it works you can find in:
http://www.voip-info.org/wiki-Convert+WAV+audio+files+for+use+in+Asteris
k
a the files .wav
chmod 755 file.wav
sox file.wav -r 8000 file.gsm resample -ql
chmod 755 file.gsm
in extensions.conf
xxxx=> xxx,x,playback(file)
Ing Javier Rios
Ing de Proyectos
04167285748
212 2637246 /2637187
-----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of
asterisk-users-request@lists.digium.com
Sent: jueves, 06 de noviembre de 2003 13:19
To: asterisk-users@lists.digium.com
Subject: Asterisk-Users digest, Vol 1 #1809 - 16 msgs
Send Asterisk-Users mailing list submissions to
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When replying, please edit your Subject line so it is more specific
than "Re: Contents of Asterisk-Users digest..."
Today's Topics:
1. Re: Questions from a total beginner (Dan)
2. CVS compile problem on asterisk (Adams, Gavin)
3. Re: Appending a # to the dialed number for
upstream carrier. (Steven Critchfield)
4. RE: MP3Player problem (David Gomillion)
5. RE: H.323 and G729: Another sad tale (G Lin)
6. Re: The Minimum Cost of Setting up an Asterisk Phone
System? (Stephen R. Besch)
7. Re: archives gsm of asterisk ??? (WipeOut)
8. Re: USB handsets/headsets?? (Nick Knight)
9. Re: archives gsm of asterisk ??? (Tilghman Lesher)
10. Re: MtSQL CDR logging (Tilghman Lesher)
11. Re: Red Alarm (Steve Underwood)
12. To SIP or Not? (David Stubbs)
13. Re: The Minimum Cost of Setting up an Asterisk
Phone System? (Steven Critchfield)
--__--__--
Message: 1
From: "Dan" <dtoma@fx.ro>
To: <asterisk-users@lists.digium.com>
Subject: Re: [Asterisk-Users] Questions from a total beginner
Date: Thu, 6 Nov 2003 18:02:49 +0200
Organization: Personal account
Reply-To: asterisk-users@lists.digium.com
Hi,
>-----Original Message-----
>From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On >Behalf Of Marrs
Seven>Sent: Sunday, November 02, 2003 10:20 PM
>>To: asterisk-users@lists.digium.com
>Subject: [Asterisk-Users] Questions from a total beginner
> ...
>I have two PC's that I want to network together using Linksys 802.11g
gear
(WRT54G ap/router & a WMP54G PCI> card in my * server). My main machine is an XP. The one I am
planning to
use for the * server has an AMD 500> processor; 64mb ram; and 30+ gb of hard drive available. I've
downloaded
the RH9 iso files to install Linux on the> proposed server. I also have one phone line coming into my home that I
would connect to the * server with a Wildcard X100P.
The minimum configuration tried for an * server was an old Compaq Armada
1700 (Celeron@300), 96MB RAM, just IP connections (3 HW phones and
several
soft phones), for test purpose only. The system was loaded at a
signifiant
level....
> 1) From what I've read, the hardware for my proposed * server is
adequate.
Is that correct? Should I put another stick of 64mb ram in the box?
Allways more memory can help.
> 4) Do the individuals at the other locations only need to obtain a
VoIP
phone and the appropriate sound card in order to gain access * at
my> location? Or is there some additional hardware/software required on
their
end of the connection? I assume that using the same VoIP
phone> at each location would be the ideal and I believe that's something we
can
do, if recommended.
The cheaper option is to use a SW phone (SIP or IAX based). You need
then
just a sound card and a headset.
> 5) My * server will be operating behind NAT on the broadband router,
but
from what I've read, that can work, although> SIP phones can have some difficulty with NAT. Can the VoIP phone used
eliminate any problems with NAT?
Use IAX... works like a charm behind NAT .
> 6) What are the pros and cons if we were to have the various locations
(individuals and eventually the second * server) communicate over a VPN?
The reliability of the connection is important. I use frequently DIAX
through a low end VPN connection (Microsoft PPTP) and no problems at
all.
Best regards,
Dan
--__--__--
Message: 2
Date: Thu, 6 Nov 2003 11:11:10 -0500
From: "Adams, Gavin" <gadams@promisant.com>
To: <asterisk-users@lists.digium.com>
Subject: [Asterisk-Users] CVS compile problem on asterisk
Reply-To: asterisk-users@lists.digium.com
Hi,
Recently I did a fresh CVS checkout of asterisk and am getting the
following errors on compile:
chan_zap.c: In function `zt_train_ec':
chan_zap.c:1078: `ZT_ECHOTRAIN' undeclared (first use in this function)
chan_zap.c:1078: (Each undeclared identifier is reported only once
chan_zap.c:1078: for each function it appears in.)
make[1]: *** [chan_zap.o] Error 1
make[1]: Leaving directory `/usr/src/asterisk/channels'
make: *** [subdirs] Error 1
[root@atlpbx01 asterisk]#
make ; make clean on Zapata, zaptel, and libpri. Currently loaded
modules are older, but I'm remiss to make install these three modules
prior to getting a clean build on asterisk too. Makes it easier to roll
back in the event of troubles.
Anyone else seeing this under RH9, kernel 2.4.20-20.9 (completely
up2date)?
Regards,
--- Gavin Adams
Promisant (Technology) Ltd.
Atlanta, GA=20
--__--__--
Message: 3
Subject: Re: [Asterisk-Users] Appending a # to the dialed number for
upstream carrier.
From: Steven Critchfield <critch@basesys.com>
To: asterisk-users@lists.digium.com
Date: Thu, 06 Nov 2003 10:22:12 -0600
Reply-To: asterisk-users@lists.digium.com
On Thu, 2003-11-06 at 00:51, Matteo Brancaleoni wrote:> hi.
>
> try with (for example)
> (XXXX is a matched 4 digit number)
>
> exten => _XXXX,1,Dial(Zap/1/${EXTEN}#,30,r)
>
> that should work.
> The idea is to put a # in the dial app.
While what Matteo suggest above is probably the best option, I'll throw
one more out that may be of use later on as your dialplan gets more
complex.
-- show application Suffix --
-= Info about application 'Suffix' =-
[Synopsis]:
Append trailing digits
[Description]:
Suffix(digits): Appends the digit string specified by digits to
the
channel's associated extension. For example, the number 555 when
suffixed
with '1212' will become 5551212. This app always returns 0, and the PBX
will
continue processing at the next priority for the *new* extension.
So, for example, if priority 3 of 555 is Suffix 1212, the next
step
executed will be priority 4 of 5551212. If you switch into an
extension
which has no first step, the PBX will treat it as though the user dialed
an
invalid extension.
> David Hindmarsh wrote:
> > Hi,
> >
> > I have a situation where our upstream carrier needs a # after we
have sent> > the dialed number.
> >
> > Is this possible.
> >
> > I have checked the Dial app and tried appending it in the exten but
this did> > not work.
> >
> > Anybody got an idea.
> >
> > Thanks in Advance
> > Dave
> >
> > _______________________________________________
> > Asterisk-Users mailing list
> > Asterisk-Users@lists.digium.com
> > http://lists.digium.com/mailman/listinfo/asterisk-users
>
>
> _______________________________________________
> Asterisk-Users mailing list
> Asterisk-Users@lists.digium.com
> http://lists.digium.com/mailman/listinfo/asterisk-users
--
Steven Critchfield <critch@basesys.com>
--__--__--
Message: 4
From: "David Gomillion" <dgomillion@eyecarenow.com>
To: asterisk-users@lists.digium.com
Subject: RE: [Asterisk-Users] MP3Player problem
Date: Thu, 6 Nov 2003 10:25:23 -0600
Reply-To: asterisk-users@lists.digium.com
Make sure you have mpg123 installed instead of mpg321... It's in the
archives somewhere... that's what fixed my install.
HTH,
David Gomillion
-----Original Message-----
From: asterisk-users-admin@lists.digium.com
[mailto:asterisk-users-admin@lists.digium.com] On Behalf Of Areski
Sent: Thursday, November 06, 2003 9:18 AM
To: Asterisk-Users Mailing-list
Subject: [Asterisk-Users] MP3Player problem
Hi all,
Is there something wrong with MP3Player ??? I always get the message
below when I try to play a MP3 :
-- Executing MP3Player("SIP/phone1-83f9",
"/var/lib/asterisk/mohmp3/02") in
new stack
WARNING[79885]: File rtp.c, Line 374 (ast_rtp_read): RTP Read error:
Resource temporarily unavailable
NOTICE[79885]: File app_mp3.c, Line 93 (timed_read): Selected timed
out/errored
out with 0
I already saw some old posts about it but no solutions !
part of an old post :
"I found that executing the new mpg123 with: mpg123 sample-hold.mp3
sometimes takes a couple of seconds to start playing. Every subsequent command
(exactly the same) starts playing immediately. Maybe this causes
the timeout in *?"
Can anyone give me a direction to solve this problem ?
Thanks in advance,
Areski
_______________________________________________
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
--__--__--
Message: 5
From: "G Lin" <glin@cosini.com>
To: <asterisk-users@lists.digium.com>
Subject: RE: [Asterisk-Users] H.323 and G729: Another sad tale
Date: Thu, 6 Nov 2003 08:33:09 -0800
Reply-To: asterisk-users@lists.digium.com
Dear all,
I just fresh CVS the asterisk code, and uncomment the G729 in the
Makefile
on asterisk/channels/h323.
I also donwload pwlib and openh323 from nufone.net/downloads, and did
following things:
1. /pwlib, make clean, make both
2. /openh323, make clean, make opt
3. /asteriks/channels/h323, make clean, make install, and it is got
error
about no chan_h323.o exists. and the make install is failed.
any one can help on this.
Thanks,
George Lin
--__--__--
Message: 6
Date: Thu, 06 Nov 2003 11:32:27 -0500
From: "Stephen R. Besch" <sbesch@acsu.buffalo.edu>
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] The Minimum Cost of Setting up an Asterisk
Phone
System?
Reply-To: asterisk-users@lists.digium.com
>Here's a cost analysis, rather crude and inspecific, of using Asterisk
>to implement a phone system. I'm really quite naive and new to all
this,>so I'd appreciate any corrections, tips, pointers, etc, from those in
>the community, who have far greater experience, knowledge, skill, etc.
>than I. Am I forgetting something important? Am I way off in my
>estimates?
>
For what it's worth, here's what we actually paid for our system:
1) CPU: Salvaged from spare parts, estimated cost to purchase:
$500
Asus A7V133, 900MHz, 256MB,60GB
2) Two Nics
$50
3) Digium T100P
$495
4) 20 GS Budgetone @$65
$1300
5) Adtran TSU600 Channel Bank (e-bay + patience)
$99
6) 2 Dual FXO plugins for TSU @$100 (used)
$200
7) 1 Dual FXS Plug in for TSU @$100 (used)
$100
8) Asterisk and Linux
$0
9) APC Smart-UPS 700 (used)
$60
Total:
$2804 $140/Phone (including phones)
$75/phone (not
including phones)
Obviously, this is a bit unrealistic for many of you, since there is no
cost for the networking infrastructure and wiring. This already existed
on our site and in fact is provided by our University. We also have a
lot of computer and electronics expertise in our lab, which really
helps. By the way, the cost was less than 1/2 the estimate for Analog
PBX hardware from an outside vendor (no installation) and less than 1/3
of the quote offered for a system with less functionality from the
University - and we would have had no control over the system and would
have been charged an addition $5/month per voice mailbox, even if though
we had to buy all the hardware! Finally, and this is a real plug for *,
one of our scientists splits his time between Buffalo and Nova Scotia. I
was able to place an extension at his Nova Scotia office for exactly
"0"
extra cost! Our University IT department wasn't even able to quote us
on that functionality.
Stephen R. Besch
--__--__--
Message: 7
Date: Thu, 06 Nov 2003 16:36:25 +0000
From: WipeOut <wipe_out@onetel.com>
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] archives gsm of asterisk ???
Reply-To: asterisk-users@lists.digium.com
Shoval Tom wrote:
>Guys, it still not working.
>
>Go here
>http://www.checkdns.net/quickcheck.aspx?domain=voip-info.org&detailed=1
>And see that it returns errors.
>
>PLEASE help.
>
>
>
None of the reported errors are critical.. They are just saying that
only one DNS server is active..
Try setting you PC up to use an alternate DNS server..
Also try "dig www.voip-info.org" on a command line and see what
results
you get..
Later..
--__--__--
Message: 8
Subject: Re: [Asterisk-Users] USB handsets/headsets??
Date: Thu, 6 Nov 2003 16:52:19 -0000
From: "Nick Knight" <nick@omniis.com>
To: <asterisk-users@lists.digium.com>
Reply-To: asterisk-users@lists.digium.com
Voipvoice handsets we tried - and are now sat on a shelve gathering
dust. The main problem was the quality of the audio - to quiet and poor
- not telephony grade for the office - perhaps good enough for home use.
=20
Just my two pennys! But still looking for a usb handset!
=20
Nick
--__--__--
Message: 9
From: Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] archives gsm of asterisk ???
Date: Thu, 6 Nov 2003 10:46:02 -0600
Reply-To: asterisk-users@lists.digium.com
On Thursday 06 November 2003 08:40, Shoval Tom wrote:> Guys, it still not working.
>
> Go here
> http://www.checkdns.net/quickcheck.aspx?domain=voip-info.org&detail
>ed=1 And see that it returns errors.
Read that page again, and you'll see that it's finding the correct IP.
Looks like it's your own DNS server which is incorrectly caching,
Tom. Might I suggest a wipe of the cache and a restart?
-Tilghman
--__--__--
Message: 10
From: Tilghman Lesher <tilghman@mail.jeffandtilghman.com>
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] MtSQL CDR logging
Date: Thu, 6 Nov 2003 10:54:01 -0600
Reply-To: asterisk-users@lists.digium.com
On Thursday 06 November 2003 05:16, WipeOut wrote:> It would appear that the "uniqueid" field is not being populated
in
> the MySQL CDR DB.. Is this an obsolete field or is a bug?
Use the source, Luke. You need to define MYSQL_LOGUNIQUEID at compile
time for it to use that field.
-Tilghman
--__--__--
Message: 11
Date: Fri, 07 Nov 2003 01:05:00 +0800
From: Steve Underwood <steveu@coppice.org>
To: asterisk-users@lists.digium.com
Subject: Re: [Asterisk-Users] Red Alarm
Reply-To: asterisk-users@lists.digium.com
Andrew Kohlsmith wrote:
>>An E1 can be a long way from the box with the right cable. However
many>>people use the wrong cable. Using a LAN cable for an E1 often gives
>>errors if the cable is more than just a few metres long. Although the
>>plugs look the same, the twisted pairs should be grouped differently
in>>an E1 cable, and it really makes a difference. If the drop cable is
only>>a couple of metres long, a LAN cable is usually adequate. This is also
>>true for T1s.
>>
>>
>
>Actually that's not entirely true.
>
>standard 568A/B wired cable does not split pairs for ethernet or DSX1
>wiring.
>
I've no idea what you mean here, since your next statements shows just
*how* they are split. :-\
>The problem is that DSX1 uses pins (1,2),(4,5) and ethernet (1,2),
>(3,6) (parenthesis show pairing). DSX1 must have the (1,2) and (4,5)
>pairs swapped to match the TX to the RX at each end, whereas normal
>
Not usually these days. The box on the wall normally needs a striaght
through cable to the card for E1s and T1s. That is why so many people
plug in a LAN cable and find it almost works.
>ethernet does not, as the switch is cross-wired. Using an ethernet
>crossover cable does not help since it is swapping (1,2) and (3,6), not
>(1,2) and (4,5).
>
Well, at least a crossover cable doesn't fool people into thinking they
got it right. :-)
>The problem with using CAT5 for long telco runs is that the impedance
is >wrong at the line clock rate (~1MHz). IIRC the impedance for telco is
>specified at 600 ohms @ 1MHz, whereas for CAT5 the impedance is
actually >
T1s are always 100-110ohm, E1s are the same when on pairs, and 75ohm on
coax. Only analogue pairs are terminated at 600ohm, and no line can
actually be greater than 120*PI (about 377) ohms - that is the impedance
of free space. Fudgy 600ohm stuff works at audio frequencies, but you
have to treat the line properly as a transmission line as the frequency
rises.
>specified at around 100MHz, where the ethernet line rate is. You can
get >away with it so long as the impedance is right, but unless you've got
the >data sheets you're playing guessing games.
>
There is no guessing involved. The impedances are pairing are all
standard. You need specs, not data speets.
Regards,
Steve
--__--__--
Message: 12
To: asterisk-users@lists.digium.com
From: David Stubbs <david.stubbs@idessa.com>
Date: Thu, 6 Nov 2003 17:13:06 +0000
Subject: [Asterisk-Users] To SIP or Not?
Reply-To: asterisk-users@lists.digium.com
Hi all,
we have go a bunch of cisco 7940 phones, i currently wondering wether
to use the sccp channel of sip. Could some one educate me on the
features / advantages of each, as I'm unsure of witch one to use?
Thanks
David,
--__--__--
Message: 13
Subject: Re: [Asterisk-Users] The Minimum Cost of Setting up an Asterisk
Phone System?
From: Steven Critchfield <critch@basesys.com>
To: asterisk-users@lists.digium.com
Date: Thu, 06 Nov 2003 11:12:56 -0600
Reply-To: asterisk-users@lists.digium.com
On Wed, 2003-11-05 at 15:03, Steve Murphy wrote:> Everyone--
>
> Here's a cost analysis, rather crude and inspecific, of using Asterisk
> to implement a phone system. I'm really quite naive and new to all
this,> so I'd appreciate any corrections, tips, pointers, etc, from those in
> the community, who have far greater experience, knowledge, skill, etc.
> than I. Am I forgetting something important? Am I way off in my
> estimates?
>
>
> The Minimum Cost of setting up an Asterisk Phone system:
>
> Fundamental Building Blocks:
>
> 1. No more phones serviced than one computer can handle.
> 2. Computer = self-built or whatever, approx. $500
> 3. OS = Linux. $0
> 4. Phones.
> Cheap Touch-tone phones: $30 each (Estimate at what I can get
> at Walmart, quantity one purchases. No digital readout, no
> programmable features.
I originally picked up this AT&T 957 phone at Office Max for $30.
Currently the links from froogle show many people offering it for around
$30, and one for $20. (http://tinyurl.com/twyi) Speaker phone, CallerID,
callerid memory, and directory dialing.
> Voip Phones $250 estimated per-phone average cost. Realize that
> costs can vary widely here!
> 5. Channel Banks. Looking at what's available on Ebay, I estimate you
> should be able to pick up a fully loaded, 24-channel FXS/FXO bank
for> $650 average. Low = $200 High = $4500. I have no way of telling
> which channel banks are compatible with asterisk. Assume that they
> are.
> 6. Digium cards: FXO card = about $100. FXS card = $125.
> 4 port FXS=$300. Prices approx. See their web site
> for exact prices.
> quad span T1 (96 lines total) $1500
> Single span T1 (24 lines) $500
> 7. Wiring. Cost of Wiring is not calculated. Assume that the premises
> is wired, with patch panels, closets, etc., already in
place.> 8. UPS, power supplies, etc: Not specifically included in the
estimates.>
> 9. Used equipment can be cheaper, but: if you need a lot, you may not
> necessarily be able to wait around for everything you need to show up
on> ebay. And what you get may not be what you wanted, etc.
>
>
> Scenarios:
> 2x8 system:
> Computer: $500
> 2 FXO : $200
> 2 4-FXS : $600
> 8 phones: $240
> --------------
> Total: : $1540 cost/phone = $192.25
This option may not work in one PC as you have more than 2 Zapata cards.
You will find it difficult to make the cards sit on their own IRQ, then
you will deal with 4 x 1000 interupts a second on the machine. Not to
mention you will be at the end of your potential expansion in a single
machine. You would find it almost the same price to switch to a T100P
and a channel bank with FXO and FXS ports. This would alleviate
headaches of IRQs, and future expansion is probably just a matter of
plugging in more phones.
> 96 line system ( FXS/FXO mix 88/8)
> Computer : $ 500
> QspanT1 : $1500
> 4 ChanBks: $2600
> 88 phones: $2640
> ----------------
> Total: : $7240 cost/phone = $82.27
>
> 192 line system (FXS/FXO mix 176/16)
> Computer : $ 500
> 2 QspT1 : $3000
> 8 ChanBks: $5200
> 176phones: $5280
> ----------------
> Total: : $13980 cost/phone = $79.43
I have a small problem with the above 2 examples in that they assume you
can get by with 11 users to a phone line. I think this is not normally
possible unless you are dealing as a telco serving residential lines
that don't have dialup internet users. I also don't feel that a $500
computer can sustain 8 T1s of traffic today. Maybe in a year the prices
will have fallen enough so that a $500 PC is adequate.
I suspect the examples above should be stated more likely as a
96 line system (FXS/FXO mix 72/T1 or PRI 3 users per line)
Computer : $ 500
QspanT1 : $1500
3 ChanBks : $1950
72 Phones : $2160
-----------------
Total: : $6110 costs/phone = $ 84.86
192 line system (FXS/FXO mix 168/T1 or PRI ~7 users per line)
Computer : $1000 Needs more power
2 QspT1 : $3000
7 ChanBks : $4550
168 phones: $5040
-----------------
Total: :$13590 costs/phone = $80.89
These also have the potential to save money month after month by having
subscribed to a T1 or PRI link and having more phone lines available for
the employees.
> Voip 24 x 192 Phones, using gnophone on existing comps & network:
> Computer : $ 500
> 1spanT1 : $ 500
> 1 ChanBnk: $ 650
> 192 gnoph: $ 0
> ----------------
> Total: : $1650 cost/phone = $10.18
>
> Voip 24 x 192 phones, using Voip Phones:
> Computer : $ 500
> 1spanT1 : $ 500
> 1 ChanBnk: $ 650
> 192 gnoph: $48000
> switch/hubs: $ ?
> ----------------
> Total: : $49650+ cost/phone = $258.59+
These also have problems. At the point you put 24 lines in a system, you
really should be getting T1 or PRI service. When you do that you remove
the channel bank from each of those examples above.
24x192 gnophones drops to a total of $1000, and cost/phone of $5.20
24x192 hardphones drops to $49000 with a cost/phone at $255.20
As has been mentioned you could get some other models of phones that go
for less than $250 each. Not to mention at the 192 mark there will be
some nice price point drops.
> Cheap $30 phones are an option, but you will not get:
> Intercom capability
So far this isn't an option with any of the asterisk systems unless you
use an overhead/loudspeaker solution.
> Message waiting capability
> CallerID type stuff.
Both of these are possible, Note features listed above regarding the ATT
957.
> Gnophones may or may not provide intercom, I haven't gotten that far
as> to find out.
I think the closest it supports would be an autoanswer.
> VOIP phones are expensive, but MAY provide intercom, and other fancy
> feature, if Asterisk will allow them. It's not clear yet to me that
> Asterisk will allow intercom. Looks like you might get message
waiting.
So far it isn't a question of whether or not asterisk will let you, but
if the VoIP protocol supports this function. So far I think we have
determined that it isn't supported but in th SCCP protocol. Maybe this
should be looked at for IAX2 inclusion and for when someone implements a
full on IAX2 hardphone. I'm betting it wouldn't be too hard to embed the
IAXIE(sp?) into a phone like the ATT 957 and have a relay trip the
speaker phone button when you wanted intercom.
--
Steven Critchfield <critch@basesys.com>
--__--__--
_______________________________________________
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
End of Asterisk-Users Digest
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Wisdom of the Ancients
