Remembering that at one point I was a hardware geek,
th GS seems to use the TMS320VC5402 PGE100 chip.
According to TI's web site
http://focus.ti.com/docs/tool/list.jthml?familyId=324toolTypeId=32
there are a number of other Codec's available, including GSM ;)
This information was figured out
101's for 68.00
http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem=3042066051category=11175
102's for 79.95
http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItemitem=3042066389category=11175
Just passing what I find along
bkw
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Asterisk-Users mailing
1. Will Asterisk route from one T1 to another perfectly? That
is, the bits that arrive on the Portmaster would need to be the
exact bits sent on the PSTN T1. Seem obvious that this should be
so.
As of this weekend it does.
2. Would you predict any trouble interfacing a Portmaster to the
On Sun, 2003-08-24 at 14:46, John Brown wrote:
Hi list,
been playing with * for a bit now and have found at
least two things that make it difficult.
1. priority numbers in extensions.conf
This line numbering gig can be a PIA. While I under-
stand there needs to be a way to branch
ct: Re: [Asterisk-Users] CLASS feature syntax
http://www.nanpa.com/number_resource_info/vsc_assignments.html
http://www.nanpa.com/number_resource_info/vsc_definitions.html
Does anybody know if there is a similar webpage for European standards ? Or
other countries standards ?
/Mike
-= On Sun, 24 Aug 2003 09:47:06 -0400 (EDT), John Vozza [EMAIL PROTECTED] said:
Just noticed that version 1.0.3.81 has been released on the
Grandstream website.
Doh. I see they still haven't decided to comply with RFC2616 (HTTP).
This has been a peeve since the first firmware release, but I
Hi!
Just wondering if any folks has had experience implementing ENUM on the
Asterisk server. If so, would you be able to provider a pointer as to how
it is done? Thanks!
Dickson
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Asterisk-Users mailing list
[EMAIL PROTECTED]
Since firmware 1.0.3.81, unless I'm imagining things, Voicemail2 seems
to be having problems.
The Grandstream and sip.conf were set to RFC2833 now with that setting I
get extra digits during Mailbox and Password phases. 222001 instead
of 2201 for example.
When both are changed to SIP info there
Am I crazy or do you have a Goto just before your Record command?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of kaku ustaad
Sent: August 25, 2003 8:33 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Why doesnt anyone reply me ?
I have posted soo
Hi,
I'm considering setting up a small call centre, but I don't want
operators to need to use their phone keypads. Supposedly, all required
call functions (dialling, answering, transfer, on hold, hang up, etc),
should be done via their scripts (be is a web interface, curses or
whatnot) and not
On Mon, 25 Aug 2003, Mark Spencer wrote:
1. Will Asterisk route from one T1 to another perfectly?
That is, the bits that arrive on the Portmaster would need to
be the exact bits sent on the PSTN T1. Seem obvious that
this should be so.
As of this weekend it does.
Nothing like
Hi,
read /usr/src/linux/Documentation/networking/netlogging.txt
regards
kapejod
--
Klaus-Peter Junghanns
CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705392
fax:+49 30 79705391
iaxtel: 1-700-157-8753
email: [EMAIL PROTECTED]
How does the portmaster distinguish between an incoming ISDN call
and incoming analog call? I know this can be done, my local ISP
can handle ISDN and analog calls on the same phone number and it
must know when the call comes in.
You could look for HDLC framing...
Whatever method the
Thanks Mark, Sorry for all the questions, I am treading on un-familiar
ground here. Can anyone think of a way to adjust how long the serial
console holds the interrupt when the kernel sends a message? Anyone know
what the purpose of this behavior is?
If we turned off the printk you'd still
You can use Agentlogin and such. If you want you can call Digium and talk
about your application to be sure we have everything you need.
Mark
On Mon, 25 Aug 2003, Miguel Bettencourt Dias (Netopia) wrote:
On Mon, 2003-08-25 at 15:09, Mark Spencer wrote:
yes, you'll need outbound spool (see
Hi,
On Mon, 2003-08-25 at 13:33, kaku ustaad wrote:
How can record a conversation with asterisk ?
I tried to use Record() but dint work for me .. here is what i tried .
http://www.loligo.com/asterisk/ has good example configurations. You'll
find a working example for recording messages there.
At 11:09 AM 8/25/2003 -0500, you wrote:
Perhaps some day there
will be a client side product/widget/whatever for Asterisk, but right now
it doesn't exist, to my knowledge that is.
I believe the Asterisk Manager will do everything Miguel wants. I will be
giving this a try in the next month or
as far as i know only the extension/context/priority (NOT
application/data side) has SetVar code. meaning you can't use
whats not there.
look in ..asterisk/pbx/pbx_spool.c line 189, notice that
ast_pbx_outgoing_app isn't passing 0-variable like line 192,
ast_pbx_outgoing_exten does.
(this was
Has anyone else had issues with upgrading from SIP 2.2 to SIP 4.4 on the
Cisco 7940? I'm following the directions outlined by Cisco. Is there a
trick that I'm missing?
Clues welcome.
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The first section basically waits for the user to dial an extension.
The second part is called if the extension they entered is invalid, then
it will go to the extension (at this point ${EXTEN} = i at priority 1
(which basically creates a loop) and the record is never called.
Try:
exten =
On Mon, 2003-08-25 at 09:09, Mark Spencer wrote:
It bugs me not having intercom/paging features. It also bugs me not
being able to look at my phones to see who's on/off.
You can do overhead paging and even stream mp3's using nbsd
Cool! Uh, what's nbsd?
--
BTEL Consulting
850-484-4535
The seller looks a little dubious. There has only been one buyer
juggsbunny. I wonder.
What about the quality of these phones? Anyone had any experience here?
- Original Message -
From: Brian West [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, August 25, 2003 12:42 AM
maybe because your email seems ot be
encoded within an attachment?
try sending plaintext!
cheers
Dave
- Original Message -
From: Armand A. Verstappen [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Monday, August 25, 2003 7:02 PM
Subject: Re: [Asterisk-Users] Why doesnt anyone reply me ?
*'s paging solution is a bad solution in light of today's phone systems.
If you need it anywhere but in a barnyard, you should plan on
selecting a different phone system. It might work in a Sams, but
certainly not in an office.
I like * as soon as I find time to get it setup, I intend to
Can someone kindly please tell me if I have to configure an
email server on my server in order for voice mail to send out the message? I am
getting my root mail filling up with returned email from Asterisk and the
messages have the voice mail messages attached. Can someone please help me?
Overhead paging is when somebody picks up an extension, speaks into the
handset, and their voice is broadcast throughout a building or zone.
What is overhead paging and how is it done with asterisk?
You can do overhead paging and even stream mp3's using nbsd
I have just received a T100P and an Adtran TSU 600 in
the mail.
I seem to be having a problem with the T100P card. So
far I have done the following:
vi zaptel.conf
fxoks=1-22
fxsks=23-24
...
vi zapata.conf
...
signalling=fxo_ks
...
channel = 1-22
...
signalling=fxs_ks
...
channel = 23-24
I
Unless you use a Valcom or Belkin solution. They make all different
types of amplifiers and zoning solutions for paging. Asterisk can work
with what ever you want hardware wise and the actual paging will sound
better than your typical phone system with paging.
Steve Lane
-Original
The Cisco SIP phones have a second voice channel available for a paging
type of implementation. Now the problem is simply of finding someone
and some time to see if it can be made to work with Asterisk.
Ray Burkholder
One Unified
519 570 0689 x2002
*'s paging solution is a bad solution in
-Original Message-
From: jerk face [mailto:[EMAIL PROTECTED]
I seem to be having a problem with the T100P card. So
far I have done the following:
vi zaptel.conf
fxoks=1-22
fxsks=23-24
...
vi zapata.conf
...
signalling=fxo_ks
...
channel = 1-22
...
signalling=fxs_ks
My zapata.conf is located in /etc/asterisk and my
zaptel.conf is located in the /etc directory.
--- Adams, Gavin [EMAIL PROTECTED] wrote:
-Original Message-
From: jerk face [mailto:[EMAIL PROTECTED]
I seem to be having a problem with the T100P card.
So
far I have done the
Ahh - I'll review over the pbx_spool.c code to see what else I can find.
I'll post any changes to the list for review.
-Original Message-
From: Richard Lyman [mailto:[EMAIL PROTECTED]
Sent: Monday, August 25, 2003 12:59 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] SetVar on
Does anyone know what this means?
WARNING[229391]: File chan_zap.c, Line 5731 (pri_dchannel): Ring
requested on channel 3 already in use on span 2. Hanging up owner.
John
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Asterisk-Users mailing list
[EMAIL PROTECTED]
Have you configured the TSU600 properly? You have to allocate each FXO/FXS
channel to a timeslot before it will work. This is not automatically done
(like the Adtran Total Access series).
Mind you, you should still have a sync light on the T1 card...
-wade
-Original Message-
From:
Hi all,
I'm not sure if this is the same thing, but if so,
let me know and I'll submit more info to bugs.digium.com.
I'm using an AudioCodes MP-104 FXO box, with snom 200
handsets:
snom200 --- asterisk --- MP104FXO --- POTS
When using rfc2833 or inband (using G.711u/a),
either way (verified
John Vozza wrote:
For what its worth, info is the only way I could ever get the GS to send
reliable DTMF even with 1.0.3.78.
I can confirm that 1.0.3.81 fixes the NTP issue I had with one phone.
However qualify=y still causes the phone to randomly lockup. (Still not
convinced this is a GS only
What version of chan_zap.c do you have ?
grep chan_zap /usr/src/asterisk/channels/CVS/Entries
regards
Martin
On Mon, 25 Aug 2003, John Congdon wrote:
Does anyone know what this means?
WARNING[229391]: File chan_zap.c, Line 5731 (pri_dchannel): Ring
requested on channel 3 already in use on
Each port is set to the proper signalling type (FXO,
FXS). I can't find any other options for the
individual ports.
As for the timing and configuration of NI, I have
tried
NI:
Timing Mode as both DTE and NI (my only choices)
Where else should I be checking?
(Before this morning, I hadn't even
Yes this is the bug, please update the ticket with what you said in email
and confirm the bug.
http://bugs.digium.com/bug_view_page.php?bug_id=171
Brenton D. Rothchild wrote:
Hi all,
I'm not sure if this is the same thing, but if so,
let me know and I'll submit more info to bugs.digium.com.
ok now lets modify that mix script to pick up on who started the monitored call and
look them up in the voicemail.conf and email it to em
Dave
[EMAIL PROTECTED] 8/25/2003 2:14:16 PM
Note that h will not be called if you park the call and pick it backup.
bkw
On Mon, 25 Aug 2003, David Harris
While trying to do a reload the command hangs. It appears
that the reload goes through successfully however the command never returns.
Started Asterisk with:
/usr/local/asterisk/sbin/asterisk
-C /usr/local/asterisk/etc/asterisk.conf
Reloading Asterisk with:
Yes... I used todays CVS.
And on another note, my system having weird issues.
Some phones would just stop working. 3-5 at a time.
A restart would fix it temporarily.
So I back tracked to my previous version 06/06/2003,
this has been my most stable version ever.
John
On Monday, August 25, 2003,
/chan_zap.c/1.90/Sat Aug 23 17:49:54 2003//
On Monday, August 25, 2003, at 03:47 PM, Martin Pycko wrote:
What version of chan_zap.c do you have ?
grep chan_zap /usr/src/asterisk/channels/CVS/Entries
regards
Martin
On Mon, 25 Aug 2003, John Congdon wrote:
Does anyone know what this means?
Anybody have any good/bad experiences with the AstDio IPS-1101 or IPH102
as a FXO gateway?
Thinking about ordering one for lab testing. Does it support US callerid?
Any other standalone inexpensive single/dual FXO gateway recommendation?
___
http://bugs.digium.com/bug_view_page.php?bug_id=156
patients grass hopper!
bkw
On Mon, 25 Aug 2003, Don Pobanz wrote:
Is it possible to configure * so that if a caller reaches voicemail for
someone in Engineering, but doesn't want to leave a message they can
press zero (0) and reach the
Has anyone else had issues with upgrading from SIP 2.2 to SIP 4.4 on the
Cisco 7940? I'm following the directions outlined by Cisco. Is there a
trick that I'm missing?
Clues welcome.
You need to upgrade to 3-03-1 first, IIRC. It's a staged upgrade,
due to some TFTP blocksize issues...
JT
The Cisco SIP phones have a second voice channel available for a paging
type of implementation. Now the problem is simply of finding someone
and some time to see if it can be made to work with Asterisk.
Ray Burkholder
One Unified
519 570 0689 x2002
*'s paging solution is a bad solution in light
Just have to get mark to put this into place.. :P
On Mon, 25 Aug 2003, John Todd wrote:
I don't see why not, after a little hacking on the Asterisk code.
ENUM is not tied to any particular VoIP protocol, and IAX would work
just as well once supported by Asterisk's ENUM lookup code.
JT
I
ah I see now.. I didn't notice that but it does atleast give you someway
to exit and go to another extension doesn't it?
On Mon, 25 Aug 2003, Brad Bergman wrote:
I certainly contemplated that very thing... but somehow it escaped
implementation.
Even as things are now, the PBX administrator
True.. thats why I say usually.. but it does yield more successful
muxing over just throwing the files together by trying to calc the in and
out diff.
bkw
On Mon, 25 Aug 2003, David Carr wrote:
If I understand the script, your technique first calculates how much longer
OUT is than IN, then
I have made a patch that uses sox instead of mpg123 to playback music on
hold. Sox, when compiled correctly will support mpg, ogg, wav, gsm and
numerous other formats. Attached is a diff file that will make change
asterisk's behavior to use sox via the perl wrapper I made.
To use this patch,
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