Wiley Siler wrote:
Hello all,
For some reason I am not showing registration in SIP.
Can anyone give me an idea what can cause this?
asterisk1*CLI> sip show registry
HostUsername Refresh State
You don't have any register => lines in sip.conf.
Maybe you are looking
Nenad Radosavljevic wrote:
Same problem here: if call come over ISDN PRI and it is for a SIP phone
that equals to strong echo situation, at the SIP end. Interestingly this
doesn't happen on all calls but it does on 95% of them. Asterisk load at
that moment is insignificant - 1 to 2 calls.
I hav
Wiley Siler wrote:
Yep. I have run "sip show peers" and things look good.
I am pretty sure I could see all my registered phones yesterday via "sip
show registry".
But then again maybe I am spacing it. I am on AAH BTW and info in AMP
and in direct check of the confs checks out fine.
"sip show regi
Anton Krall wrote:
Guys, this is weird.. Today I started having some problems with calls been
dropped. Im suing X100p cards (clones) and I have this setting on my zatala
fle:
[channels]
[snip]
busydetect=yes
busycount=4
Can the echotraining be messing things? Do I need to enable callprogress or
so
[EMAIL PROTECTED] wrote:
I upgraded a DVG-1120M to a DVG-1120S. Everything works great, but I'm
having some caller ID issues on incoming calls sent to the SIP device.
Using debug on the device, the caller ID looks fine - just as I set it in
Asterisk. However, the phone is showing "CID TRANSMISSI
I have no idea. I live in the USA so I don't normally need busydetect.
Anton Krall wrote:
Why does busydetect actually drop calls while stile talking?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling
Sent: Viernes, 11 de Marzo de 2005
David Uzzell wrote:
I have seen the list of codecs for the ATA 186's but not sure if it was
100% or not.
I want to know really is it possible to run GSM or ilbc on them or is a
G729 lic the only way to get a low bandwidth codec?
This is the list of codecs that I have seen.
RxCodec and TxCodec—C
On March 13, 2005 09:57 am, Nigel Burgess wrote:
[door]
exten => s,1,Dial (SIP31,15)
exten => s,2,Playtones(dtmf)
However the call hangsup before trying to play the DTMF tone.
When a Dial happens, the dialplan stops until the call is
disconnected. See "show application dial" to see how you can se
Peter Svensson wrote:
On Sun, 13 Mar 2005, Robert Hajime Lanning wrote:
There are SMS sending gateways out there, but they are sending
only, no way to receive. This is fixed in the IM solution by
giving the "system" an account of its own.
Whatever gave you that idea? Most operators have an inter
Robert Hajime Lanning wrote:
Well, as far as I know there is no such service in the USA. Take in
mind that SMS is not so popular in the states, email is, and every
cell phone in the US that I have seen that supports SMS, supports SMS
to email from the phone as well.
um, backwards. E-Mail to SMS.
Robert Hajime Lanning wrote:
Robert Hajime Lanning wrote:
um, backwards. E-Mail to SMS. I have not seen the other way
around.
Both Cingular and Verizon supports both.
I have not tried this, nor have I seen any documentation mentioning
it. Do you or anyone else have a pointer for the info?
Espe
Raoul Bönisch wrote:
Hello!
I'd like to Flash() a modem line (BRI) with Asterisk. It is a
passive ISDN-card connected to a hardware PBX. I use ISDN4Linux.
I recognised that unfortunately the Flash() application flashes
Zap devices only. Now I am wondering how I could flash Modem/ttyI0.
The source c
Colin Anderson wrote:
I'm going to do a deployment of LocustWorld MeshBoxes in some of our remote
locations. Build 90 comes with Asterisk 1.0, and our plan is to use the
MeshBoxes as a WAP for non-Asterisk uses but also to add a 2nd NIC to deploy
Snom's in the remote location. This works fine (was
Brian McCrary wrote:
Hello,
I'm trying to run an ATA behind a NAT device, and am confused on exactly
what the qualify config option does, other than send NOTIFY packets.
Outbound calls work fine, but inbound calls do not go through. With
qualify=yes and nat=yes, my show sip peers looks like:
77
Roman Zhovtulya wrote:
Hello,
I wonder if I would have to sacrifice anything if I set "NAT=yes" for
all sip clients I have, regardless of whether they are behind the NAT or
not.
The idea is to have the setting that works regardless of whether the
user is behind the NAT or not, since I'm not sure wh
Raoul Bönisch wrote:
* Eric Wieling <[EMAIL PROTECTED]> [2005-03-14 16:56]:
Raoul Bönisch wrote:
Flash is an analog thing. It does not even apply to ISDN.
So how does the "R" key on my ISDN-telephone work then?
I suspect it sends an ISDN specific "put call on hold&quo
Ron Joffe wrote:
Hey folks
I have a new setup with a TDM400P for a pair of analog extensions and a few
SIP phones. We seem to be experiencing a bunch of "Crackeling" when talking
between the analog and SIP extensions.
Any ideas?
Yes. Check the suggestions given to the other guy that posted this
César Davi Ávila do Nascimento wrote:
Talk about skype is forbidden, but to be impolite is allowed...
Great list!
Skype does not interface with Asterisk in any way whatsoever. You
could just as well have asked if someone knows what RNA sequence 42 in
the turnip genome is for. About as many peop
[EMAIL PROTECTED] wrote:
Hello,
I upgraded my office from Asterisk 1.0.0 to Asterisk
CVS-HEAD-03/13/05-13:14:04 this weekend, and are now
experiencing some problems accessing voicemail. The web based interface
works fine, in addition to dialing 8500,
which is mapped to:
exten => 8500,1,Voicemail
Kanuri, Seshu (Company IT) wrote:
I am using Asterisk 1.06 Stable.
When I dial my Mobile Number to check Voice Mail or my Bank Account
Phone Access Number, the IVR System on the other end asks me to enter
*2378 to transfer to an attendant.
But When I press *2378, Asterisk tells me that it cannot tr
Kanuri, Seshu (Company IT) wrote:
Thanks for the pointers. Here is my Features.conf where I have tried my
best to use Asterisk to give away control. I have enabled ## as the
combination key for Asterisk (in quick succession) to retain control,
but otherwise ignore the key presses.
I don't run CVS-H
Neil A. Hillard wrote:
Using X-Lite to dial extension 400, I hear it ring and then get answered
and I hear about 0.1 of a second of the on hold music and then silence.
If I use the 'line 1' button to put the call on hold and then take it
off again I hear another 0.1 of a second of the music. This
David Zanetti wrote:
I've been beating my head a bit against the 1.0.6 Debian builds of
Asterisk, using an E100P (E1, single span) board.
In machines I've built in the past (back in 1.0.0 days), config I'm
using and that card and 1.0.0 driver combo worked fine.
ztcfg reports no problems:
SP
Once you run Dial from an AGI script, you lose control of the call via
the AGI script.
Jean-Michel Hiver wrote:
(obviously if you do other magic in your dialplan this needs to be
adjusted. The important part is the 'g' flag to Dial (go on after
hangup), and the NoOp which echos the dialstatus
Richard J. Sears wrote:
Hey Everyone,
I am using NuFone for 866 inbound service and I am trying to figure out
the callerid part of it. Any call into my * system just shows "Toll Free
Call" and will not give me the calling party's caller ID info.
Is this just something I have to live with using NuFO
Jason Becker wrote:
Daniel Burget wrote:
I have read every thread, I have Redhat 9, Asterisk 1.0.6, 2 T1 lines
connected via TE405P. Everything works great, except MOH. I added an
exten with MusicOnHold(30), and it plays just fine. Conferences have
music when no one is in. I have SIP phones. When I
John Goerzen wrote:
Over at http://www.voip-info.org/wiki-Asterisk+tips+911, I see these
extensions.conf lines:
exten => s,1,SetVar(SET_EMERG_FLAG=0)
exten => s,n(checkavail),ChanIsAvail(${EMERGENCY_TRUNK})
exten => s,n,SetGlobalVar(EMERGENCY=1)
exten => s,n,SetVar(SET_EMERG_FLAG=1)
exten => s,n(di
Thomas Andrews wrote:
If I comment out the following line in zapata.conf I would expect
asterisk to "forget" the cli information for that channel when I reload:
callerid="Uniden Dead" <(256) 428-6125>
... but it doesn't; I have to restart asterisk for it to take effect.
The funny thing is that the
Anton Krall wrote:
What do you think?
CPU0
0: 16148159 XT-PIC timer
1: 4 XT-PIC keyboard
2: 0 XT-PIC cascade
5: 0 XT-PIC usb-uhci
8: 1 XT-PIC rtc
10: 161351663 XT-PIC usb-uhci, w
Asterisk is not a SIP proxy.
Wei Su wrote:
We encouter a situation where we need to use SIP info to convey infomation
for one end point to another endpoint. I use asterisk to do the test and
find asterisk does not forward the SIP info to another endpoint, but act as
UAS and returns a 4xx error mess
C F wrote:
Now consider this (this works with the cisco 7960, even if you put a
7914 with it, it will still use all 20+ plus buttons this way, if CW
is disabled on the phone):
exten => 123,1,Dial(SIP/${EXTEN},30,tr)
exten => 123,2,Voicemail(u${EXTEN})
exten => 123,3,Playback(goodbye)
exten => 123,4
cmisip wrote:
No path to translate from SIP/fwdpulvercom-dd5a(2) to Phone/phone0(1)
I don't know why the above message is printing codec numnbers, rather
than names. *shrug*
"show codecs" will tell you what codec number are what codec name.
It appears that your Phone/phone0 is using G723.1. Look
Tyler wrote:
I think you're looking for the 'ChanSpy' application that seems to have
inexplicably vanished from the asterisk CVS..
http://www.voip-info.org/tiki-index.php?page=Asterisk%20cmd%20ChanSpy
If anyone has any info on this, let me know as I'm in a similar
situation.
As far as I know ChanSp
Kevin P. Fleming wrote:
Matt Gibson wrote:
This is what I'm sending from my dhcpd server.
option ntp-servers 10.x.x.x;
option tftp-server-name "ftp.x.x.x";
option time-offset -18000;
Keep in mind that using TFTP for a Polycom boot server is sub-optimal,
because you have to renam
Brian McCrary wrote:
Hello,
I'm wondering if anyone has benn able to successfully get g726-16
passthrouhg to work? I am wanting to use this codec instead of g729 as
I'm running out of DSPs using a high complexity codec on the Ciscos. I
would think it would work just as g729 does, which has been w
It means the caller hung up in the middle of the voicemail app.
Anton Krall wrote:
So far, nobody has been able to tell us what this error means.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of MF Hulber
Sent: Lunes, 21 de Marzo de 2005 02:54 a.m.
To: Ast
Alessandra Grasso wrote:
My objective is to estimate the performances of *
How much the trancoded can influence the performances?
Thanks,
show translation recalc 30
--
Always do right. This will gratify some people and astonish the rest.
Mark Twain
___
A
Sys Admin wrote:
I am setting up a new asterisk based call center. I just read:
http://www.voip-info.org/wiki-IAX+versus+SIP
After reading this and other google results for "IAX vs SIP" is there
any reason why i should use SIP anywhere !!
Because most equipment doesn't support IAX
--
Always do righ
Tom wrote:
This is what I have suspected all along is that the signaling and timing
constraints on the PRI are such that you basically need asterisk running as a
real-time process. The whole point of the thread (in my mind) is if there is
anyway to cause X to not run as such a real-time process so
Neill Wilkinson wrote:
All,
newbie to Asterisk and just trying to get a load of bits together
including PSTN interface using Digium Quad E1 interfaces using EuroISDN.
Question can I/how do I get access to the ISDN reason codes for call
disconnect?
/path/to/asterisk/docs/README.variables
Pay spec
Yes. http://www.fnords.org/~eric/asterisk/downloads/macros.inc Pay
special attention to the [macro-dial-result]
joachim wrote:
Did someone actually got this to work ?
The examples on voip-info are not correct i think.
It will never go to priority 2, since the call already got hungup at
that po
exten => _91NXXNXX,1,Dial(${PSTN}/${EXTEN:1})
exten => _91NXXNXX,2,Macro(dial-result)
joachim wrote:
Thnx Eric,
Could you also post your extensions.conf ? (or just the extension
calling that macro ?)
Joachim
At 04:48 22/10/2004, you wrote:
Yes. http://www.fnords.org/~eric/asterisk/downlo
Yes it works. It will go to priority 2 if the call was NOT ANSWERED for
any reason (busy, number not in service, etc). You may need to add ,,g
on the Dial line to get Asterisk to go to priority two if the CALLEE
hangs up.
I do not do post call processing if the CALLER hangs up.
joachim wrote:
Kevin Walsh wrote:
Benjamin on Asterisk Mailing Lists [EMAIL PROTECTED] wrote:
On Sat, 23 Oct 2004 02:16:44 -0700, Trevor Peirce <[EMAIL PROTECTED]> wrote:
G723 GSM ULAW ALAW G726 ADPCM SLINR LPC10 G729A SPEEX ILBC
G723 - - - - - - - - - -
Brian J. Rathman wrote:
I am having several users complain about not being able to use the # button when dialing into IVR's, etc, because the # key prompts for transfering the call to another extension. Is there a way to still provide transfer capability, but not use the # key? I am using SNOM 200
IAX does not correctly set the HANGUPCAUSE for a LOT of things. Look at
DIALSTATUS or look at the dial-result macro on
http://www.fnords.org/~eric/asterisk/downloads/macros.inc
Matt Schulte wrote:
I am connecting Asterisk to Asterisk to PSTN (Either by SIP or PRI) and
am having some issues deal
*I* consider it a bug. Mark (if I recall correctly) considers it "just
the way it works".
Matt Schulte wrote:
Interesting, would this be considered a bug or is it rather intentional?
Or is that a dumb question ;-)
-Original Message-----
From: Eric Wieling [mailto:[EMAIL PROTEC
Andrew Thompson wrote:
[EMAIL PROTECTED] wrote:
Hello everyone!
Version 1.0.2 is now available for Asterisk, Zaptel, and libpri.
Would it be possible to get official changelogs for these releases?
(If they're out there, please point me toward them.)
Thanks.
General changelog is part of the downlo
Delete the modules that are new to CVS-HEAD when you migrate to CVS v1-0
STABLE
[EMAIL PROTECTED] wrote:
I just downloaded it and tried to run it on a development machine and
now it shows the following on startup
[app_realtime.so]Oct 26 09:35:46 WARNING[-154464]: loader.c:248
ast_load_resourc
1.0.0 had G726 as well
Brian West wrote:
The 1.0.2 release has g726.
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Adam Greenbaum
Sent: Tuesday, October 26, 2004 11:06 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] G.726
Hi,
I am
Steve Underwood wrote:
Kevin Walsh wrote:
Brian McSpadden [EMAIL PROTECTED] wrote:
I would buy a $10 license of Digium's g.729 and do some testing with
that. From what people have told me, that "open source" g.729
implementation causes crashes, performs more poorly, and just isn't a
good idea. T
George Gardiner wrote:
I would be grateful for any pointers in the right direction. In short, I get CallerID to display on Xten and a SipTone II; but have failed miserably to get my BudgeTone 101 to display anything other than the phone's own number.
The BT101 can only display callerid number. It
Leah Newmark wrote:
> Oct 26 13:49:01 WARNING[638995]: file.c:473 ast_openstream: File
letters/d
does not exist in any format
Oct 26 13:49:01 WARNING[638995]: file.c:761 ast_streamfile: Unable to open
letters/d (format GSM): No such file or directory
Run "make datafiles" in the Asterisk source d
Me wrote:
Is it possible to transfer a caller to another internal extension with a
plain analog phone attached to an ATA?
Use the FLASH button or hangup the phone for 1 second, the dial the
number to transfer to, then hang up.
begin:vcard
fn:Eric Wileing
n:Wileing;Eric
email;internet:[EMAIL PROTEC
Mandrake specific: urpmi openssl-devel
Thomas Hupfeldt wrote:
- Original Message -
From: "Brian West" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
<[EMAIL PROTECTED]>
Sent: Thursday, October 28, 2004 2:17 AM
Subject: RE: [Asterisk-Users] where do i fi
Alessio Focardi wrote:
Hi all !
I played around for a few hours with a polycom 500 phone and it seems me that the dtmf
mode is not configurable, looks like it only has inband mode.
While this is ok with G711 I assume that will result in some troubles
using G729, altought I cant test it because I ha
Unless something has changed since the last time this was discussed on
the mailing list the answer is "No, call deflection does not work on PRI".
Craig Foley wrote:
(Sorry if you've seen this already on the mailing list, but I've not
seen this post coming up for me, even though my mailing setting
That will, of course HIDE any BUSY or telco messages. And the caller
will never know of they dialed an invalid number or of the number they
dialed is busy. Do you think he REALLY wants that?
Steve Totaro wrote:
add an r to the end to your dial statement
- Original Message - From: "Step
appens with
(hopefully with a note where the ringback actually stops).
Steve Totaro wrote:
then what is REALLY the solution?
- Original Message - From: "Eric Wieling" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
<[EMAIL
Rich Adamson wrote:
I'm trying to config a temp iax connection between two current * boxes.
One is behind a firewall, the other uses a registered IP.
I config'ed the * box behind the firewall to 'register' with the one that
has a registered IP. The registration is occuring and the CLI indicates:
-
Rich Adamson wrote:
Rich Adamson wrote:
I'm trying to config a temp iax connection between two current * boxes.
One is behind a firewall, the other uses a registered IP.
I config'ed the * box behind the firewall to 'register' with the one that
has a registered IP. The registration is occuring and t
Cirelle Enterprises wrote:
In a recent upgrade to version * 1.0.2 I have noticed
a new behavior in the Zapateller() function.
It now produces the 3 tones you get when you hear
the "were sorry" message from the phone company.
Anybody notice this "New" feature?
SIT aka Special Information Tone is the
Robin van Leyden wrote:
Does any body have any information about Dialogic MSI board workink with
asterisk.
According to this document the MSI model is not supported:
http://www.asteriskpbx.org/index.php?menu=hardware
Keep in mind that the Dialogic drivers for Asterisk are closed source
and cost
Michael Rowley wrote:
Hello,
Trying to rewrite my dialplan, and it is a little complex. But my
extensions.conf redirection works, but the referred to contexts result
in "invalid extension" Please help... I have the extension set to 's'
currently, but originally it was 6044. The change didn't
John Lange wrote:
Asterisk seems to have a problem with the Cisco 7905. If the user is on
the phone with another call, asterisk reports:
-- Executing Dial("SIP/206.XX.XXX.XXX-08f899d8", "SIP/204X83|10") in new stack
-- Called 2044X83
-- Got SIP response 488 "Not Acceptable Here"
Tim Thompson wrote:
Centrex is a type of line and I do not believe there is a compatible card
for *.
FXS isn't going to cut it. Centrex is a digital type line.
That is just plain wrong.
Centrex services can be delivered via a variery of ways. In my
experience it's usually via analog FXO lines, b
Benjamin on Asterisk Mailing Lists wrote:
If you refer to the urban legend that IAX always needs a server to
stay in the media path, then you would be wrong. IAX has a mechanism
that for all practical purposes is equivalent to a SIP reinvite
through which the end points then transition to a mode by
Patrick wrote:
Hi all,
I have an unavailable/busy voicemail message recorded which plays fine.
After the message has been played to the caller, vm-intro.gsm is also
played ("Please leave your message after the tone. When done hang up or
press the pound key"). I do not want this vm-intro.gsm to be
Benjamin on Asterisk Mailing Lists wrote:
On Wed, 3 Nov 2004 11:57:44 -0600, [EMAIL PROTECTED]
<[EMAIL PROTECTED]> wrote:
Ok, check out ftp.digium.com/pub/asterisk/g729/unsupported/linuxppc
Knock yourself out (oh, and look at the date on that file too :-) )
date?
"The requested URL /pub/asterisk/g
Damon Estep wrote:
> I also tried RepsonseTimeout but then you can only have one timeout
event (ie hangup). Sure would be nice to specify progressive timeout,
like on first timeout goto, on second timeout hangup.
I assume loligo/silence/10 is an empty gsm file that I have not
installed...
[auto-at
Nicklas Bondesson wrote:
I don't hear a busy-tone when calling an external extension that's busy.
I just get the Busy Here 486 message in the debugging log. Any ideas?
What do you hear?
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digiu
Nicklas Bondesson wrote:
I don't hear anything. There's no sound at all.
Nicklas
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling
Sent: den 7 november 2004 17:21
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re:
Dan wrote:
Hi,
I have upgraded the Asterisk Server after several months and now there
is an issue with the CallerID Name information.
When I call from DIAX (IAX2) to ATA186(SIP) I get the correct CallerID
name/number.
When I call from ATA186(SIP) to DIAX(IAX2) I get the correct number, but
the C
x.xxx.xxx
-- SIP/xxx.xx-7d58 is busy
== Everyone is busy/congested at this time
Thanks
Nicklas
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Eric Wieling
Sent: den 7 november 2004 20:02
To: Asterisk Users Mailing List - Non-Commercial Discussion
Su
Damon Estep wrote:
http://www.voip-info.org/wiki-Asterisk+bounty+outcall+notification+appli
cation
Starting at $500 from me, if you have interest in the application please
add to the bounty and lets get this application written!
This is not a real application, but this is how I do voicemail outcall
Damon Estep wrote:
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Eric Wieling
Sent: Sunday, November 07, 2004 8:39 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] New bounty for voicemail
Nicklas Bondesson wrote:
Just like this? It doesn't seem to work though.
[wx3trunk-outgoing]
include => internal-sip-callers
exten => _X.,1,Dial(SIP/[EMAIL PROTECTED],,T)
exten => _X.,101,Busy
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://l
Paul Rodan wrote:
Nobody responded so I’m sending this out again. I need help on stopping
the “Change caller ID on forward” trick that either Cisco or Asterisk
keeps doing. My upstream provider doesn’t like it.
This doesn't help?
'f' -- Forces callerid to be set as the extension of the lin
Nathan C. Smith wrote:
What would one enter to get the stable or 1.x version of Asterisk and
associated modules via CVS? I've googled and wikkied but I'm using the
wrong terms or asking the wrong questions.
Generally, you follow the directions.
http://www.asteriskpbx.org/index.php?menu=download
_
Nicklas Bondesson wrote:
There's still one problem though. It's seems like the _X.,102,Busy picks up
all events such as 404 user not found - if you dial an extension that
doesn't exist.
Any clues?
No. Dial either goes to the next priority when the Dial command times
out or current priority+101
Mike Coakley wrote:
Hello All,
I originally send this message a little while ago. I realize that if anyone
had any input they would have commented but maybe... Just maybe someone that
could have commented missed it so I am resending it.
I also figured (since I forgot it the first time) the version
Nick Cobley wrote:
I have a need to connect up asterisk to an Exicom GSX 418/816, this
will be a very simple setup, just one extension on the Asterisk box so
only one line to the PABX required.
Problem lies in the Exicom being a Key system and and we cannot source
any Single Line Modules for this s
xed out or
not able to provide the 2 wire extensions, but in a lot of cases they
seem to have CO lines spare.
But just wondering if there is actually any technical advantage to
hooking it up this way?
On Wed, 10 Nov 2004 21:58:41 -0600, Eric Wieling <[EMAIL PROTECTED]> wrote:
Nick Cobley wro
el Flynn wrote:
Brian West wrote:
Ok I have known about this for over a month now and have yet to see
anything
come of it. Just a little note that Sysmaster is packaging up
Asterisk in
their product and not giving a notice with the product with an offer
to the
source of the the GPL software the
You CANNOT download Cisco firmware without a CCO account AND support
contract.
Jerry Geis wrote:
Did you search for "7912 sip software" in the search tab?
That is where I found mine.
Jerry
Hello
I did register, but I only find manuals and guides. But no software.
And when I go to Downloads-Voi
Paradise Dove wrote:
how can i get a CCO account?
or is there any other place for cisco downloadable stuff without user/pass?
or a free to all CCO account!!!??
A CCO account is free. However you cannot download any software unless
you have a Cisco Support Contract. You cannot LEGALLY download th
Joe Greco wrote:
Really? Wouldn't it be nice, then, if Digium explicitly stated that this
was their intention, in their little agreements?
[snip]
That'd be the FSF calling this both "ethically tainted" and showing a loss
of "moral standing". I'd be happy to put it to them to see if there is a
mor
Paul Fielding wrote:
I've currently got Asterisk configured to take incoming calls and send
them directly to my voicemail. I'd prefer to keep this approach rather
than sending people to a menu first.
What I want to be able to do is have voicemail come up, but if someone
presses a key, such
Paul Fielding wrote:
I've currently configured incoming calls to simultaneously ring an
analog phone (via TDM400P) and two SIP phones. I'd like to have it
also simultaneously dial out the TDM400P on a PSTN to ring my cell
phone, and have the first one to answer win the battle.
In my diggi
Scott Stingel wrote:
Brian-
I was quoted (verbally) something on the order of $60 per month for a single
BRI by SBC in San Francisco about 60 days ago. I thought that was high..
It has been a while, but the last BRI I ordered from BellSouth/Louisiana
was about US$109/month.
__
Dinesh Nair wrote:
On 16/11/2004 00:08 kido noagbodji said the following:
i have an easy way to install the codec under FreeBSD? It was tough
enough
to install asterisk even with the FreeBSD ports.
i do not believe that digium sells the g729 codecs for freebsd. however,
i too am a freebsd user,
Other than the standard codec issues? No. disallow=all and allow=ulaw
in [general] in sip.conf. NO other allow= lines.
Chris TenHarmsel wrote:
No one?
On Mon, 15 Nov 2004 12:40:42 -0500, Chris TenHarmsel <[EMAIL PROTECTED]>
wrote:
Hi all,
I've attached the output from asterisk with "set verbo
Régis MARTIN wrote:
Hi,
I know the question was already asked but I never found an answer to
this problem. So, I try again… (things changes J)
Is there a way to play a specific message or sound from the start during
the dial command.
I want to do exactly the same thing that the “m” option of
Henry Devito wrote:
Hi, I have looked through the sounds files, but I think I am
overlooking the file that speaks Comidian Mail. I would like to change
this to a more friendly greeting, like “Thank you for calling Comedian
Mail” and instead of it saying Mailbox have it say enter your mailbox
Matt Riddell wrote:
Lex Lethol wrote:
Does anyone know if this needs any special modification to work
outside the US? I have setup my country's correct tone info and
tested thru the indication.conf file.
Question would be, where does my zaptel device get the tones expected
for the busydetect proce
I think at this point it would be a good idea to contact Digiun's tech
support. I've never seen a card not generating interrupts. Digium does
provide install support for their cards, which is what you need right now.
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[E
Henry Devito wrote:
How do I implement the dial by name application on my IVR? I searched
through the archives with goggle and can’t find the basic how to
information.
"show application directory" will show you.
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Huddleston, Robert wrote:
Is directory included w/ Asterisk or external app... I'm running older
release * so j/c
-Original Message-
From: Eric Wieling [mailto:[EMAIL PROTECTED]
Sent: Tuesday, November 16, 2004 2:52 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Su
Angel Diaz wrote:
Hi all,
Does somebody know what's new with SS7 and * ?
I'm very interested. Is it ready ? I'm prepared to pay if necessary.
Join the asterisk-ss7 mailing list at http://lists.digium.com/
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Joe Greco wrote:
Who to generate "ring tone" to a calling party when the call is passed
to an extension.
The asterisk answers correctly, plays welcome message and ring an
extension, but the caller does not here the rings.
Did you tell Asterisk to indicate ringing?
Asterisk will ALWAYS indicate rin
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