[EMAIL PROTECTED] wrote:
Ron,
It is a multi-reported problem, yet no resolution.
I would suggest it is a bug. I have had intermittent
success with POTS provided by AllTel in Texas.
My opinion, you're SOL and there is very little you can do.
I keep hoping that someone at digium will pick up
Hello,
Does anybody know if Asterisk can support QSIG protocols to be
interconnected with a Traditionnal PABX?
(Using a HFC chipset based ISDN card to emulate NT Interface)
Thank you in advance ;-)
Ignace
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[EMAIL
Hello,
Does anybody know if Asterisk can support QSIG protocols to be
interconnected with a Traditionnal PABX?
(Using a HFC chipset based ISDN card to emulate NT Interface)
Thank you in advance
Ignace
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Asterisk-Users mailing list
[EMAIL
I hope this question isn't flamebait. I don't know anything about voice T1.
What are the tradeoffs in terms of asterisk's design and performance
whether traffic is handled by one type or the other?
I wonder about the economics, too.
Thanks.
B.
___
What version of the Phone firmware are you running ? I had the same problem
until I upgrade to
1.0.4.54
Chris
- Original Message -
From: pesb [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Wednesday, March 24, 2004 9:41 PM
Subject: Re: [Asterisk-Users] Re: Asterisk SIP + Grandstream
Do your php support GD ?
You can simply check it with a phpinfo !
More info about gd (configuration, installation) :
http://www.php.net/image
On Wed, 2004-03-24 at 21:12, Robert Boardman wrote:
Hi I'm trying to install but I think I have a problem!!!
Would I be correct in saying if I
Yes php sysinfo say gd is complied inb
any other clues?
Robb
Areski [EMAIL PROTECTED] said:
Do your php support GD ?
You can simply check it with a phpinfo !
More info about gd (configuration, installation) :
http://www.php.net/image
On Wed, 2004-03-24 at 21:12, Robert Boardman
Hello,
A few weeks I wrote a message about a program to manage faxes.
A few people responds which I appreciate.
For this program I made a project site on http://www.sourceforge.net.
The project page for the fax program is: http://tafm.sourceforge.net
If you would like to download the program
Thanks for your feedback. We will look into it.
Sincerely,
Stephen Karrington
Dreamtime.net Inc.
http://www.dreamtime.net
http://www.emailblaster.us
Corporate Office
101 California Street, 22nd Floor
San Francisco, CA 94111-5802
Voice - 877-203-9308
Fax - 310-943-2606
Dreamtime is your
Well ...
For starters, in your sip.conf you have
dtmfmode=rfc2833
but your phone setup gives
send_dtmf=in-audio
In your post (below) you also left out
authenticate_password=gol
but that may be an oversight?
BTW: My GS setup uses dtmfmode=info (in my sip.conf for each
phone)
and send_dtmf=SIP_IPNFO
This should be taken off the list
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Hermann
Wecke
Sent: Wednesday, March 24, 2004 8:37 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] IAX2 International Termination
On Thu, 25 Mar 2004, Anton Tinchev
Can you try:
http://yourdomain/asterisk-stat/graph_stat.php?min_call=fromstatsday_sday=25days_compare=2fromstatsmonth_sday=2004-03
And tell me about the result !
-Areski
On Thu, 2004-03-25 at 11:13, Robert Boardman wrote:
Yes php sysinfo say gd is complied inb
any other clues?
Robb
Hi Areski
it comes back with a blank page?
Robb
Areski [EMAIL PROTECTED] said:
Can you try:
http://yourdomain/asterisk-stat/graph_stat.php?min_call=fromstatsday_sday=25days_compare=2fromstatsmonth_sday=2004-03
And tell me about the result !
-Areski
On Thu, 2004-03-25 at 11:13, Robert
I have the following problem in playback:
When any sound file is played back, it is garbled for a few
seconds and the following error displays:
Sched_settime: Request to schedule in the past?
After about 5 seconds, the sound clears up and the error
stops.
What gives???
Download asterisk-addons from the CVS. Compile it the same way you
compile asterisk and it's other modules. Make sure you have MySQL
installed and running. Then read the file
/usr/src/asterisk-addons/doc/cdr_mysql.txt for information on how to
create the necessary tables in your database. The
I've got a single inbound analogue line setup with 2 phone numbers and
distinctive ring and I'm trying to setup distinctive ring detection to
separate calls and put a distinctive ring to the extensions based on
what number was called...
Problem is it seems most countries send a distinctive
Sounds like it s missing smth in your php conf !
have a look to this good tutorial
http://www.zend.com/zend/tut/tutsweat3.php
Check your configuration with the conf information provided there
and then try to make working a jpgraph sample on your server...
Hope that it will help,
Regards,Areski
Dear Chris,
My firmware version is 1.0.4.39, how can I make the upgrade?
where (url site) can I get the firmware?
thanks again,
Pablo S.
On Thursday 25 March 2004 06:32, Chris Stenton wrote:
What version of the Phone firmware are you running ? I had the same problem
until I
On Thu, 2004-03-25 at 12:50, pesb wrote:
My firmware version is 1.0.4.39, how can I make the upgrade?
where (url site) can I get the firmware?
http://www.grandstream.com/BETATEST/
--
Dave Cotton [EMAIL PROTECTED]
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Necessary to create a register in the Asterisk, more it has that to send the
information:
username, password, sip proxy, outboundproxy, domain/real.
Help to decide this problem me?
Thank´s
Joao Carlos Moura
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I have tried to connect asterisk (which I use through hisax isdn4linux
device) with mediatrix sip device with g729 codec
asterisk can not connect with mediatrix (it connects when ulaw/alaw are
used) when g729 is forced
any ides what to do?
Sip read:
SIP/2.0 200 OK
Call-ID: [EMAIL PROTECTED]
Hey all,
Thought I'd share a curiosity I found when trying to use heartbeat
software for asterisk failover (this may already be common knowledge to
some/many, but I hadn't seen mention of it yet). The default ha-linux
ip-takeover script uses ifconfig to create an ethernet alias to which a
UNSUBSCRIBE
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[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users
Duane,
On Thu, 25 Mar 2004, Duane wrote:
Problem is it seems most countries send a distinctive ring then the
caller ID, however here it appears a short ~50ms ring is sent, followed
by a pause with caller ID *then* the proper ring/distinctive ring is
sent, is there any simple way to get
Seems to me this thread should be taken off-list. It's effectively a beta
test list for a Dreamtime product, so Dreamtime should set up their own list
and at most send a single invite to it to the asterisk list.
Carey
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Asterisk-Users mailing list
Try calling application Hangup at the ends of the extension chains.
Works for me.
Bob
[EMAIL PROTECTED] wrote:
Ron,
It is a multi-reported problem, yet no resolution.
I would suggest it is a bug. I have had intermittent
success with POTS provided by AllTel in Texas.
My opinion, you're SOL
You need a G729 license for asterisk to make a connection. You have to get
them from diguim, they are $10 a channel. They do give you a single channel
demo license, you just have to get it from them.
Wes
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Marko
Hello,
Thats a good idea. We will go and set up our own list for this
discussion. Thanks for the suggestion.
Sincerely,
Stephen Karrington
Dreamtime.net Inc.
http://www.dreamtime.net
http://www.emailblaster.us
Corporate Office
101 California Street, 22nd Floor
San Francisco, CA 94111-5802
Sorry about the post to the wrong level of the thread, but something was
wrong with the first copy of the message (i.e., my mail reader wouldn't
display it). Comments are inline.
I tried Stephen advice and it did not work. I stil got the 404 error
[general]
dtmfmode=rfc2833
This does not match
We thought about this
option. I guess the IAX2 is not the problem. We believe the real problem will
be the user interface.
snom would have no
problem providing the platform (hardware plus operating system and stuff like
audio), but we simply dont want to open another development branch
I have a problem w/ a IAX termination provider. Recently when people would
call over IAX, they could hear everyting even dial an extension. When the
extension picked up, I could hear them but they could not hear me. My ZAP
device works fine, it's just coming in over the IAX. I updated to the
I've successfully installed Asterisk and have Microsoft's Instant
Messenger connecting. We can make VoIP calls between clients without a
problem, however we cannot send text instant messages between clients.
From what I can tell this should be possible using IETF SIMPLE or RFC 3428
(SIP Message
Humble apologies for using list space for this. The message is
actually for Stephen Karrington.
I wrote a lengthy reply to you directly (Stephen), but it was bounced
by your spam filter. If you are interested in seeing it, please
contact me directly, and let me know how else to forward that
Hi all,
My SoftFAX is now up to spandsp-0.0.1h.tar.gz. This version has a number
of changes in the way the V.29 modem works. It also has some missing
functionality in the T.30 implementation filled in - it was not handling
EOM messages.
The previous version failed for several reasons with a
I've successfully installed Asterisk 0.7.2, before we used 0.4.0 than it was
working well but we needed context with date/time. In 0.7.2, we have trouble
when use DTMF. We make outgoing calls from CISCO 7960 and use Cisco 2621
like gateway. About audio/voice all is working right, but DTMF don´t
- Original Message -
From:
Christian
Stredicke
To: [EMAIL PROTECTED]
Sent: Thursday, March 25, 2004 10:05
AM
Subject: RE: [Asterisk-Users] IAX and
Snom200
We thought about
this option. I guess the IAX2 is not the problem. We believe the real
UNSUBSCRIBE
No. I don't want to.
--
_/ _/ _/_/_/_/ _/_/ _/_/_/ _/_/
_/_/_/ _/_/ _/_/_/_/_/ _/ K e v i n W a l s h
_/ _/_/ _/ _/ _/_/ _/_/[EMAIL PROTECTED]
_/ _/ _/_/_/_/ _/_/_/_/ _/_/
Hi all,
I try to install a G.729 license in SCSI system with a IDE CDROM
but I can't do it. Any one has experience to do this?
Regards,
srsergio
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[EMAIL PROTECTED]
Sergio,
Did you try to install G729 while you had a CD in the CDROM drive?
Regards,
Christopher J. Wolff, VP CIO
Broadband Laboratories, Inc.
http://www.bblabs.com
--__--__--
Message: 4
From: Sergio Serrano [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Date: Thu, 25 Mar 2004 17:48:21 +0100
Subject:
Sergio Serrano wrote:
Hi all,
I try to install a G.729 license in SCSI system with a IDE CDROM but
I can't do it. Any one has experience to do this?
Regards,
srsergio
Here is the wiki page for g729:
http://www.voip-info.org/wiki-Asterisk+G.729+Licensing
It's not
Juan == Juan J Sierralta P [EMAIL PROTECTED] writes:
Juan I been playing with RxFax ... I received a FAX and it seems
Juan that the aspect ratio of the image is different, ... The image
Juan resolution is 1728x1092.
Traditional fax has two resolutions: 98 lines/inch and 196 lines/inch.
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Wednesday 24 March 2004 10:51 pm, Adam Hart wrote:
Comment below...
Steve wrote:
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Wednesday 24 March 2004 08:45 pm, James H. Thompson wrote:
No guarantee then when public IPs match that
Yes I have mounted CDROM first with automount(/dev/cdrom) and second
manually(/dev/hde) but nothing.
Any idea?
srsergio
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Andrew
Thompson
Enviado el: jueves, 25 de marzo de 2004 17:59
Para: [EMAIL PROTECTED]
Hal A. Lightwood wrote:
I've successfully installed Asterisk and have Microsoft's Instant
Messenger connecting. We can make VoIP calls between clients without a
problem, however we cannot send text instant messages between clients.
From what I can tell this should be possible using IETF SIMPLE
If memory servers, and everyone feel free to flame away if it serves
badly, the library only searches hda,hdb,hdc, and hdd. Try switching
where your controller is, that may solve it.
Derek
-Original Message-
From: Sergio Serrano [mailto:[EMAIL PROTECTED]
Sent: Thursday, March 25, 2004
Hi, Im new in Asterisk world.
Could Somebody tell me if Asterisk solution is cpable
to give IP-IP PBX service ?
If the answer is yes, wich is the moodule o service
name?
Thanks
Mariano
__
Do you Yahoo!?
Yahoo! Finance Tax Center - File online. File on time.
Certainly there is the NAT issue and this should not be underestimated.
Also IAX allows optimisation of existing bandwidth between Asterisk
servers.
The
SNOM guys should look over their shoulders at Verbiage who are bringing an IAX
phone to market. I suspect it will have a lot of interest
Here's a recap of what I am hearing:
1) Everybody (thus far) is in favor of trying to standards-track (or at
least do an Information RFC) on IAX2.
2) IAX2 needs to have AES encryption added prior to submission.
3) IAX2 needs to have non pin-wheeling NAT support added (i.e. support for
Dear
i have two box and i want made some stress test with one TE410P and a E100P
with only one span 1
Server
TE410P
Span1-- PBX
Span2---E100P Box
The Box with TE410P is Mandrake 9.2 with P4 HT
#zaptel.conf
span=1,0,0,ccs,hdb3,crc4,yellow
span=2,0,0,ccs,hdb3,crc4,yellow
Hi there !
any hint about this error that I got connecting an eads matra pbx to
asterisk with a zaptel pri interface ?
my cfgs:
zaptel.conf
loadzone = us
defaultzone = us
span=1,0,0,ccs,hdb3,crc4
bchan=1-15
dchan=16
bchan=17-31
zapata.conf
[channels]
pridialplan=unknown
signalling=pri_net
Thanks for the quick, if not very detailed answer. Obviously I am
interested in this capability, is there some reason we couldn't work on
this? I believe SER might support it (it seems to work between FWD
clients at least), why not asterisk? What would be required to implement
this
--Original Message Text---
From: Barry Fawthrop
Date: Thu, 25 Mar 2004 11:07:24 -0500
- Original Message -
From: Christian Stredicke
To: [EMAIL PROTECTED]
Sent: Thursday, March 25, 2004 10:05 AM
Subject: RE: [Asterisk-Users] IAX and Snom200
We thought about this option. I guess the
May problem is: I need to know when and between which channels
connection is setup and hungup. Is there a way to learn this from
manager interface? There are Link/Unlink events but then appear more
than once during single connection, ie while calling from IAX to SIP I get:
Event: Link
Channel1:
I've finally uploaded the newest (LARGE) list of sound clips in .gsm
format to the bugtracker.
Please see http://bugs.digium.com/bug_view_page.php?bug_id=985
for details and a full sound file list (and a tarball of the sounds
in gsm format.) AIF soundfiles are available if you really,
All,
I have some odd message waiting issues with a variety of my SIP clients.
Each client has an entry like this in sip.conf;
[2200]
type=friend
host=dynamic
context=intern
username=2200
secret=2200
dtmfmode=rfc2833
mailbox=2200
As you can see, I specify which a mailbox. This works fine on my
Looks like you have to have one side of the direct connection supply a clock
source. Try having box 2 source the clock on that span:
span=1,1,0,ccs,hdb3,crc4,yellow
Also, I've never used the Yellow option, so I don't know how that effects
things.
But anyway, I've done exactly what you want to
On Wed, Mar 24, 2004 at 06:25:01PM -0600, Michael Graves wrote:
My wife has a business as well as I. I had envisioned the three line
comming in to * in such a manner that her line would be handled in a
different manner than my two.
Sure. Just change the context for that Zap channel, and have
Oh, one more thing:
You must use an E1 crossover cable when you directly connect one E1 to
another (not using a PBX). You can make one yourself, as follows:
TEST CABLE WIRING-
It's easiest to cut up a standard ethernet CAT5 cable and rewire the
connections. Only 4 wires are needed. Please use
Received my Pulver WiSIP phone a couple days ago. Has anyone successfully
gotten the phone to work with 128-bit WEP? I've tried entering the key via
the keyboard (ugh), turning off WEP then adding the key via the web
browser (minor ugh), and all steps in between.
The only thing that may be an
John Todd wrote:
Please see http://bugs.digium.com/bug_view_page.php?bug_id=985
for details and a full sound file list (and a tarball of the sounds
in gsm format.) AIF soundfiles are available if you really, really
want them, but they're huge and I don't feel like putting them in the
It looks like we have to create an Internet Draft which is assigned to the
relevant working group for revision, questions, comments, more revision,
then it may or may not become an RFC. Unfortunately, the IPTel working
group appears to be made up of people who are heavily invested in SIP.
That's
Hal A. Lightwood wrote:
Thanks for the quick, if not very detailed answer. Obviously I am
interested in this capability, is there some reason we couldn't work on
this? I believe SER might support it (it seems to work between FWD
clients at least), why not asterisk? What would be required to
On Thu, 25 Mar 2004 13:51:15 -0500, Tim Sailer wrote:
On Wed, Mar 24, 2004 at 06:25:01PM -0600, Michael Graves wrote:
My wife has a business as well as I. I had envisioned the three line
comming in to * in such a manner that her line would be handled in a
different manner than my two.
Sure.
Title: RE: [Asterisk-Users] SoftFAX/spandsp
Hi,
This is to confirm that with spandsp-0.0.1h Dialogic VFX/40ESC
faxing started working, a great deal for us! Thank you, Steve.
Will test more ...
There is a downside though - looks like this release causes
page cutoff. We've had it before - 2
I am trying to use Asterisk as a pure voicemail system and have the following
setup:
I have the * setup as a SIP peer to a softswitch. When someone calls a number on
the softswitch and no one picks up the phone, the softswitch forwards the call
to the * using SIP. The message header of the SIP
Hi guys,
I am a newbie and having problem to enter a conference room.
Here is an extract of my config files:
# extensions.conf
; Or a conference room (you'll need to edit meetme.conf to enable this room)
;
exten = 8600,1,Meetme,1234
# meetme.conf
[rooms]
;
; Usage is
NAT traversal is a huge issue I agree with Michael and Brian
what with the latest viruses etc... security is and will be
more and more of an important issue, many SOHO and small corps.
Often don't have the know how or finanical backing to implement
standard/conventional security and internet
Lal, Deepak (Contractor) wrote:
I am trying to use Asterisk as a pure voicemail system and have the following
setup:
I have the * setup as a SIP peer to a softswitch. When someone calls a number on
the softswitch and no one picks up the phone, the softswitch forwards the call
to the * using SIP.
Try doing an answer first:
exten = 8600,1,Answer
exten = 8600,2,Meetme,1234
Might also be worth doing a Meetme(1234) instead of Meetme,1234. I
believe both should work, but..
-Original Message-
From: Mailling LIst [mailto:[EMAIL PROTECTED]
Sent: Thursday, March 25, 2004 3:17 PM
Mailling LIst wrote:
Hi guys,
I am a newbie and having problem to enter a conference room. Here is an
extract of my config files:
I had a look on the mailing list archive but did not find anything
regarding this problem. Thanks in advance for your help
This is really a FAQ. You need a
Hi Gavin,
Works OK with my 128-Bit WAP.
Remove the Space or put in an underscore and try again.
Regards
Dave
-Original Message-
Gavin Adams wrote: -
Received my Pulver WiSIP phone a couple days ago. Has anyone successfully
gotten the phone to work with 128-bit WEP? I've tried
The ${RDNIS} variable in the dialplan would contain that information.
${RDNIS} for SIP is in CVS HEAD. A patch for 0.7.2 is at
http://www.fnords.org/~eric/asterisk/downloads/
On Thu, 2004-03-25 at 14:07, Lal, Deepak (Contractor) wrote:
I am trying to use Asterisk as a pure voicemail system and
A quick search of Yahoo found quite a few reports of issues in various
devices with spaces in the SSID. Seems a lot of implementations fail to
properly handle the space. Definitely sounds like a WiSIP issue, but
might be worth removing the space from your SSID if at all
convenient
Sean
Dialing in from the pstn to sip phones (x-lite softphone on winders and
a grandstream handytone-286 ata), I hear the sip phone ring a few times,
I ran into the same thing with Cisco 7960. Looks like the logic in the
sip channel has changed recently.
Add a ,r to the end of your Dial statements in
1. Aastra 390 to Digium tdm.
2. E1 line. Redirected DID number to USA. tried with several phones -
simemens s45 gsm, panasonic, ge.
Everything works fine, but dtnf relaying is broken.
Stephen Karrington wrote:
Thanks for the feedback. What kind of phone are you using?
Sincerely,
Stephen
Will do guys. It didn't even occur to me until I was heading into the
office. WiSIP + beer == FATAL_USER_ERROR!
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Sean Cheesman
Sent: Thursday, March 25, 2004 3:53 PM
To: [EMAIL PROTECTED]
On Thu, Mar 25, 2004 at 01:58:54PM -0600, Michael Graves wrote:
On Thu, 25 Mar 2004 13:51:15 -0500, Tim Sailer wrote:
On Wed, Mar 24, 2004 at 06:25:01PM -0600, Michael Graves wrote:
My wife has a business as well as I. I had envisioned the three line
comming in to * in such a manner that
I was following the development of chan_sccp on the Lambda website,
but sometime last week all of the links went dead, bugs, cvs, etc.
Did the development move?
Dan
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[EMAIL PROTECTED]
To all;
I've got two installations of asterisk. The last one (installed a few
days ago) is from the FreeBSD ports, and many thanks, because it
compiled BEAUTIFULLY! However, I can't run it. Everytime I start
asterisk, I get a segmentation fault. asterisk -c reveals :
[...snip...]
I think Asterisk should
have no problem with NAT, even when used with SIP. I mean just listen for the
first RTP packet and send the stream where it comes from (thats called symmetrical
NAT). I think everybody is doing it like this now and they are selling their
stuff for thousands and
- Original Message -
From: Adams, Gavin [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Thursday, March 25, 2004 1:08 PM
Subject: [Asterisk-Users] Semi OT: WiSIP and WEP
The only thing that may be an issue is that my SSID has a space in it
Test WAP. When I view it the first time on the
Hi all,
I am able to track incoming h323 calls with phone number by using
amaFlags=billing or amaFlags=documentation. But is it possible to tracking
the incoming IP at the same time?
If I would like to restrict incoming h323 access to certain IP, should it be
done on asterisk or oh323 level?
Lal, Deepak (Contractor) [EMAIL PROTECTED] wrote:
I am trying to use Asterisk as a pure voicemail system and have the
following setup: I have the * setup as a SIP peer to a softswitch. When
someone calls a number on the softswitch and no one picks up the phone,
the softswitch forwards the call
Here's the main problem I've run into. I'm trying to use FWD with
Asterisk, and am behind a nat device (dsl modem with nat built-in, no way
to bind the IP directly to a server/PC). I also have a SIP gateway, a
Welltech 3502 (it goes by many other names, always see it with the 3502
model number).
Hi,
We experienced a problem this week on our asterisk box (ast-2) that has
a T1 coming in and talks over IAX2 to a second Asterisk box (ast-1). In
the current setup we use ast-2 for outgoing phone-calls only, it takes
calls (over IAX2) from ast-1 and routes those calls out over the T1.
This
Tested from Belgium
Very good quality, sometimes breaking up a little.
The phone I used is a Snom200 behind *, gsm codec.
Ping times are 110 - 115 ms.
Did not try dtmf sending.
Robert Sprockeels
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[EMAIL PROTECTED]
Excellent work Steve.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steve
Underwood
Sent: Thursday, March 25, 2004 10:34 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] SoftFAX/spandsp
Hi all,
My SoftFAX is now up to spandsp-0.0.1h.tar.gz. This
I see that I can purchase G.729 licenses for my Asterisk server, but I
have seen that many phones support a G.729 variant like A or B. Are these
suppoted by the same G.729 codec in Asterisk?
--
Carlos Chavez
Computer Engineer, CCNA
Corporativo Lacer S.A. de C.V.
si funciona con el A y B
Miguel Cavazos
On Thu, 2004-03-25 at 22:47, Carlos Chavez wrote:
I see that I can purchase G.729 licenses for my Asterisk server, but I
have seen that many phones support a G.729 variant like A or B. Are these
suppoted by the same G.729 codec in Asterisk?
--
Carlos Chavez wrote:
I see that I can purchase G.729 licenses for my Asterisk server, but I
have seen that many phones support a G.729 variant like A or B. Are these
suppoted by the same G.729 codec in Asterisk?
B is just the fixed point version of A (from memory) - so it works the
same
No. The FXO on the 182 is only usable from the box itself. It's for
calling local numbers.
Erick Weber V. wrote:
Hi to everyone:
Does someone know if the ATA 182 works OK with asterisk or should I get a
HandyTone 486 instade or an ATA 186 and a FXS to FXO converter
Thanks
Erick
On Thu, 2004-03-25 at 09:33, Steve Underwood wrote:
exten = 5678,1,txfax(/tmp/testfax.tif|caller)
There are a zillion fax and tiff formats. I'm trying to figure out what
output format I should tell GhostScript to use. Any suggestions on
which format to try?
These are the formats GhostScript
When I start or reload * I always get this error (once).
Can someone point me in the right direction to fix this.
WARNING[1150495040]: chan_sip.c:495 retrans_pkt: Maximum retries exceeded on
call [EMAIL PROTECTED] for seqno 102 (request)
Simon
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What kind of specs do I need for a asterisk box that will have a pri for
pstn and about 65 sip phones
I was thinking a Xeon 3.05
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First, I am very new to this software. If I missed a searchable archive, please point
me in the right direction.
I am wishing to know if Asterisk can be used to do a Call Drop scenario.
This is where someone calls, Asterisk answers, ask for the number that the person
wishes to dial, gets the
I've put up a new dev version of Firefly
(http://www.virbiage.com/firefly/download/firefly-dev.exe)
Notable Changes:
DTMF now works with SIP
Speex codec has been added
1 crash bug fixed - 2 more to go (if you can crash Firefly, send me the
Hex address - probably stored in event viewer under
I have three * servers that all talk to each just fine, and
all talk to other * servers (like NuFone, VoicePulse, etc.).
I have hard-phones connected to Sipura SPA-2000s on two of
the * servers via a local network connection. The third *
server only gets connected to remotely, both with IAX and
When you use firefly in SIP mode it does not un-register with * on exiting
the software
Simon
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Adam Hart
Sent: Friday, 26 March 2004 11:48
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] New minor release
Hi Eric,
I was all day trying and came up with this:
gs -q -sDEVICE=tiffg3 -sPAPERSIZE=a4 -r204x196 \
-dNOPAUSE -sOutputFile=$TIFFILE -- $PSFILE
I'm using a modified version of salsafax/sambafax to enable a
print2fax option for windows/linux clients.
You add a printer to cups and share it via
Hi List,
Two boxes
A has a PRI
B terminates SIP devices
A --IAX-- B
Both on the same switch, same IP network.
Call from PSTN to A gets pushed via IAX to B - Sip device
with no problems.
Call from Sip device - B via IAX - A - PSTN
will drop exactly 5 seconds after the call is
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