Greg Kedrovsky wrote:
I have a TDM40B, 4-port fxs card. Each port seems to have it's own
little board on the fxs card. Each little board is not sodered in, but
rather "hangs" (I have a vertical case for the server) on what I would
call jumper pins (sorry, I'm not a profession geek, just a wannabe)
> Hi,
>
> Please excuse me if my question seems too simplistic. I have been reading
> the mailing list for some time and I am still a bit confused. Here is the
> scenario that I would need to achieve and am wondering if asterisk is the
> correct software to use.
>
> (h323) (h323/SIP
Andy Powell wrote:
lo,
Is there a single central location for code and applications other than CVS? I'm talking about code that can't/wont be included in CVS for various reasons? Does the wiki have this sort of thing? I've done some code for the Cepstral TTS engine (bkw has done some updates too)
The Meetme needs to monitor DTMF and be able to trigger an AGI.
and
When the Meetme room is emptied it needs to be notified back to * so you
can trigger a clean-up event or what not.
that is what i would like to see. :)
There really aren't any. Once you're in a conference, you can only
exit t
On Wednesday 04 February 2004 08:58, Brian West wrote:
> Search bugs.digium.com their was a patch for seconds but I don't
> think it was applied yet
It was applied; it's just not part of the default. 'S' is the digit
for speaking seconds.
-Tilghman
__
> Nufone is setup for it and it works great
IIRC you need to request it; I had trunking on and could only hear people;
they couldn't hear me.. Turned it off and it's all good.
Regards,
Andrew
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MLS Drop for SysAdmin wrote:
an associate of mine sent me an email of the slick sheet on this one. I
understand that mentioning this vendor has resulted in some
flamethrowing on the list, and I do not want to cause trouble - just
looking for some info.
Thanks!
Sam Z
I have one in service. It
Hi all,
I've been trying on and off again for several months to get my 7960
(MGCP 5.3) working with * with no success. As you know, working MGCP
configs for non-ATA Ciscos seem to be very hard to come by. I'm not
shooting for the moon here, just trying to get dialtone at the moment.
The prob
Hi all,
I've been trying on and off again for several months to get my 7960
(MGCP 5.3) working with * with no success. As you know, working MGCP
configs for non-ATA Ciscos seem to be very hard to come by. I'm not
shooting for the moon here, just trying to get dialtone at the moment.
The prob
Hi,
Please excuse me if my question seems too simplistic. I have been reading
the mailing list for some time and I am still a bit confused. Here is the
scenario that I would need to achieve and am wondering if asterisk is the
correct software to use.
(h323) (h323/SIP) (h
Would be great if I could actually download it. It looks nice, does it
work?
On Wed, 2004-02-04 at 08:37, Walker Haddock wrote:
> The link to the download site for the softphone is:
> http://www.lipz4.com/lipz4.htm
>
> On Wed, Feb 04, 2004 at 08:57:00AM +, WipeOut wrote:
> > An article I cam
Bob Klepfer wrote:
voicemail is misspelled - would that do it?
Yup that fixed it, thanks for all the help
Regards
Chris
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> Voicepulse told me that there was no additional charge to enable trunking.
GASP & SWOON!!! You received a response out of voicepulse?
bkw
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Andy's code and my code are the same code basically. I cleaned up a few
things and added the noanswer option. Other than that Andy did all of the
hard work.
bkw
On Wed, 4 Feb 2004, Brian Capouch wrote:
> [EMAIL PROTECTED] wrote:
> > Feedback for the list. I compiled Andy's code. Installation
Matteo Brancaleoni wrote:
hi.
I've gs working under NAT,
simply put nat=yes into sip.conf section if *,
then enable nat into the gs, without any stun server.
If I do that (which I already have tested) * will try to initiate rtp
with dest IP eq the inside adress ie 192.168.0.160.
BTW nat=yes
Nufone is setup for it and it works great
On Wed, 4 Feb 2004, Stephen R.
Besch wrote:
> Chris Clifton wrote:
> > The majority of sip to pstn gateway providers (vonage, voicepulse, and
> > others) appear to be setup for a one line only type of set up. Their web
> > sites seem to be heavily geare
> I have a TDM40B, 4-port fxs card. Each port seems to have
> it's own little board on the fxs card. Each little board is
> not sodered in, but rather "hangs" (I have a vertical case
> for the server) on what I would call jumper pins (sorry, I'm
> not a profession geek, just a wannabe). One of
Chris Clifton wrote:
The majority of sip to pstn gateway providers (vonage, voicepulse, and
others) appear to be setup for a one line only type of set up. Their web
sites seem to be heavily geared for these one line setups.
Anyone willing to comment on what type of pricing plans these providers
off
On Wed, Feb 04, 2004 at 09:21:28AM -0600, Mark Spencer wrote:
> Asterisk 0.7.2 is now released and contains lots and lots of bug fixes
> from the bug tracker. Highly recommended for people running 0.7.1.
Great, I was going to grab the latest CVS version after business hours today
anyway :)
Cheer
When I access voicemail remotely, from a gsm phone say, some extra
characters get inserted in my dtmf tones: when I type , *
understands 88f8f8 (it always seems to be 'f'):
-- Incorrect password '88f8f8' for user '2130' (context = )
And the 'f' always starts after the second digit. Might it b
On Wed, 4 Feb 2004, John Todd wrote:
> At 10:18 AM +0100 2/4/04, Andy Powell wrote:
> >
> >lo,
> >
> >Is there a single central location for code and applications other
> >than CVS? I'm talking about code that can't/wont be included in CVS
> >for various reasons? Does the wiki have this sort of
> On Wed, Feb 04, 2004 at 07:47:08AM -0700, Greg Hill wrote:
>>
>> Automotive parts places sell products like lok-tite (a thread locker
>> compound for mechanical fasteners).
nail polish and liquid-paper work fine for this sort of stuff.
>
> Thanks. I'll give that a try.
>
> -Greg
>
> --
> Mutt
John Todd wrote:
> At 10:18 AM +0100 2/4/04, Andy Powell wrote:
>>
>> lo,
>>
>> Is there a single central location for code and applications other
>> than CVS? I'm talking about code that can't/wont be included in CVS
>> for various reasons? Does the wiki have this sort of thing? I've
>> done som
Is there a way to get the extension that was used to park the call using the
ParkAndAnnounce command into a variable? Or a variable that is set? I
would like to create an application that allows the person the call is being
announce to be able to accept the call (by pressing 1) or send the call t
Chris Lee wrote:
I am having problems with my dial plan, please help me locate the
problem:
In the following dialplan, I am not able to press 8 to get to
voicemail main while the 3000 mailbox unavailable message is being
read in the background.
What am I doing wrong?
[well-road]
;includes
in
At 10:18 AM +0100 2/4/04, Andy Powell wrote:
lo,
Is there a single central location for code and applications other
than CVS? I'm talking about code that can't/wont be included in CVS
for various reasons? Does the wiki have this sort of thing? I've
done some code for the Cepstral TTS engine (bk
Asterisk 0.7.2 is now released and contains lots and lots of bug fixes
from the bug tracker. Highly recommended for people running 0.7.1.
Mark
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Hi,
I have setup * from IAX2 and for the client the IAXphone (sokol).
When I try to call an demo-extension there is a notice:
[114696]: chan_iax2.c:4341 socket_read: rejected connect attempt from my_ip
Any idea ?
Regards,
MarinBlu
Do you Yahoo!?
Yahoo! SiteBuilder - Free web site building
Do you evern include the [well-road] context?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Chris Lee
Sent: Wednesday, February 04, 2004 9:11 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Whats wrong with dialplan?
I am having problems with my dia
[EMAIL PROTECTED] wrote:
Feedback for the list. I compiled Andy's code. Installation went well
(except for me misspellng something in the dialplan) with no problems.
The Application works great. Will run down Brian's and give it a try too.
Hope you can do us a "HOWTO."
Cepstral would be a majo
Hi,
> -Original Message-
> In the following dialplan, I am not able to press 8 to get to
> voicemail main while the 3000 mailbox unavailable message is
> being read in the background.
> What am I doing wrong?
You need to put the exten = > 8,... Line in the same context as extensions,
On Wed, Feb 04, 2004 at 07:47:08AM -0700, Greg Hill wrote:
>
> Automotive parts places sell products like lok-tite (a thread locker
> compound for mechanical fasteners).
Thanks. I'll give that a try.
-Greg
--
Mutt 1.4.1i on Slackware 9.1 Linux
Curridabat, San Jose, Costa Rica
http://www.greg-a
> Has anyone else experienced this problem? What could I do to solve it
> (seat the little card a little more permanently)?
Haven't experienced it but I would think that a small bead of silicone
sealant would hold things in place. As the stuff cures it will release
acetic acid but so long as yo
I am having problems with my dial plan, please help me locate the problem:
In the following dialplan, I am not able to press 8 to get to voicemail
main while the 3000 mailbox unavailable message is being read in the
background.
What am I doing wrong?
[globals]
;physical-phones
p1 = SIP/p3000
p2
Feedback for the list. I compiled Andy's code. Installation went well
(except for me misspellng something in the dialplan) with no problems.
The Application works great. Will run down Brian's and give it a try too.
Robert
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Search bugs.digium.com their was a patch for seconds but I don't think it
was applied yet
bkw
On Wed, 4 Feb 2004, Deepakumar JV wrote:
> Thanks for your reply Brian.
>
> I am able to get only the hour and minute but not the seconds. I need
> seconds also, any suggestions?
>
> Regards
> Deepak
>
On Wed, Feb 04, 2004 at 08:42:04AM -0600, Mark Spencer wrote:
> Usually the cards seat pretty well. Do you have a green or blue
> TDM40B card?
Blue.
--
Mutt 1.4.1i on Slackware 9.1 Linux
Curridabat, San Jose, Costa Rica
http://www.greg-and-sue.com/screenshot.jpg
Yahoo Instant Messenger ID: gre
On Wed, 4 Feb 2004, Greg Kedrovsky wrote:
> Has anyone else experienced this problem? What could I do to solve it
> (seat the little card a little more permanently)?
Automotive parts places sell products like lok-tite (a thread locker
compound for mechanical fasteners). A drop or two of that, plac
Thanks for your reply Brian.
I am able to get only the hour and minute but not the seconds. I need
seconds also, any suggestions?
Regards
Deepak
- Original Message -
From: "Brian West" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, February 04, 2004 02:23 PM
Subject: Re: [A
Usually the cards seat pretty well. Do you have a green or blue
TDM40B card?
Mark
On Wed, 4 Feb 2004, Greg Kedrovsky wrote:
> I have a TDM40B, 4-port fxs card. Each port seems to have it's own
> little board on the fxs card. Each little board is not sodered in, but
> rather "hangs" (I have a ve
The link to the download site for the softphone is:
http://www.lipz4.com/lipz4.htm
On Wed, Feb 04, 2004 at 08:57:00AM +, WipeOut wrote:
> An article I came across this morning..
>
> http://www.itnews.com.au/storycontent.asp?ID=12&Art_ID=18128
--
DataCrest, Inc. -- Technically Supe
Absolutely no argument from me on that front, hands down the Cisco 7940/7960
are a damn good IP phone, and compared to the existing Norstar handsets we
have, a far better phone overall.
The handsfree functionality on the Cisco's is truly awesome, the mic pickup
and clarity is far better than the
SayUnixTime will do that just give it the format you want.
SayUnixTime([unixtime][|[timezone][|format]])
unixtime: time, in seconds since Jan 1, 1970. May be negative.
defaults to now.
timezone: timezone, see /usr/share/zoneinfo for a list.
defaults to machine defa
I agree. app_cepstral is a damn fine app and has been banished to the
edges of the earth because the theta engine isn't open src. I even added
a standalone build for app_cepstral... so you can download it.. make it
and install it without much trouble. :(
Andy maybe we can go thru and pickout th
Hello
I am looking for a AGI application that
can say the current time with seconds, but i don't need the
day/year.
Has anyone got this already?
Thanks in advance
Deepak
hehe ya I have to admit they are very featureful. :P Asterisk is still a
baby i'm sure sip phones will get better with time. But you do have to
admit that the cisco 7960's are damn good phones.
bkw
On Thu, 5 Feb 2004, Christopher Lee wrote:
> Then you've got to hand it to Nortel, they do know
Then you've got to hand it to Nortel, they do know how to make a damn good
phone extensions for lazy people like me :-)
I actually believe this isn't the case with the Nortel Meridian systems, as
I noticed when using one it wouldn't accept the numbers without first
pressing that extensions DN key
I have a TDM40B, 4-port fxs card. Each port seems to have it's own
little board on the fxs card. Each little board is not sodered in, but
rather "hangs" (I have a vertical case for the server) on what I would
call jumper pins (sorry, I'm not a profession geek, just a wannabe). One
of my little boar
I have one word for you... LAZY!
bkw
On Wed, 4 Feb 2004, Christopher Lee wrote:
> Out of interest, does anyone know if it's possible to get the 7960 to start
> accepting a number while on-hook, without having to press "NewCall", the
> line button, or speaker button?
>
> This is just something I
One friend with Cisco 7960 with public IP address connect to my
* box and I called me to my home phone through a X100P.
He can hear me clearly and I cannot hear him.
I thought the problem could be a NAT in the middle.. but there is no NAT.
Any thoughts?
Isamar
_
Question.. is the 7960 on the same subnet as your asterisk server? I have
a 7960 registered with 3 diffrent asterisk servers. All 6 lines. Running
6.1 and has 12 days of uptime.
bkw
On Wed, 4 Feb 2004, John Todd wrote:
>
> So, I've managed to consistently lock up my Cisco 7960 (SIP 6.1) to
>
> Out of interest, does anyone know if it's possible to get the 7960 to start
> accepting a number while on-hook, without having to press "NewCall", the
> line button, or speaker button?
>
> This is just something I was used to with the Norstar extensions, I could
> immediately start dialing the
Rich,
If the Mediatrix uses the Caller-ID field to select which channel to
use, then you really have no
choice but to do that. As you pointed out, the Caller-ID info is not
(and cannot) be passed to
the PSTN line.
Rich Adamson wrote:
Possible Mediatrix 1204 fxo sip gateway workaround
Need so
Out of interest, does anyone know if it's possible to get the 7960 to start
accepting a number while on-hook, without having to press "NewCall", the
line button, or speaker button?
This is just something I was used to with the Norstar extensions, I could
immediately start dialing the numbers for
> What are my support options for CALEA with Asterisk?
>
Don't think you'll find any support without some major AGI/Manager programming,
and then not likely to interface with the calea requesters.
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On Wed, Feb 04, 2004 at 12:38:37AM -0500, William Suffill said:
> I will be moving in a few months and I'm concerned as to what kind of
> bandwidth I would need to work effectively. The reason I posed the
> question here is simple most of my work is remote SSH to various
> BSD/Linux machines but a
What firmware and sip versions are you using? I have several Polycom phones
on my system right now and I've never had any registration problems with
them.
Instead of leaving the host as dynamic try declaring an IP address(that's
the only difference I see between your sip.conf and mine).
If you a
When I make a call and the other party is busy I do not hear anything
but a free ringing phone.
Also, if I call a call center with a voice menu the phone keep ringing
without any sign of life.
I tried early b3 with these ones but nothing change much:
http://www.voip-info.org/tiki-pagehistory.php
Hi,
- Original Message -
From: "Grzegorz Nosek" <[EMAIL PROTECTED]>
> ...
> A packet dump should reveal all.
>
> What do you think?
Good idea.
Use Debug feature in DIAX and send both phones and asterisk logs to check
them offline (you'll have timestamps on each phone log so you can rebui
On Wed, 4 Feb 2004 11:13:52 +0200, Dan wrote
> Hi,
>
> From: "Peer Oliver schmidt" <[EMAIL PROTECTED]>
>
> > >>I have 4569 opened and forwarded/NATed to my *. I am on the same
network
> > >>as the * server, a friend is remote. After about a minute you
loose the
> > >>connection.
> >
> > > This is
G'day,
On Wed, 4 Feb 2004, Senad Jordanovic wrote:
> I have two cards in one of the servers. If I bind SIP port to public IP,
> it all works fine. If I do not bind to specific IP (ie. Bind = 0.0.0.0),
> I get segmentation fault while starting *.
I have this configuration also, with two network c
I have two cards in one of the servers. If I bind SIP port to public IP,
it all works fine. If I do not bind to specific IP (ie. Bind = 0.0.0.0),
I get segmentation fault while starting *.
Can SIP (and other protocols), bind to more then one IP address?
If yes, what is syntax?
SJ
Andy,
I would be interested in your Cepstral engine code.
Regards,
Robert
Friedrichshafen, Germany
Andy Powell said:
> lo,
>
> Is there a single central location for code and applications other than
> CVS? I'm talking about code that can't/wont be included in CVS for various
> reasons? Does the wi
Andy Powell wrote:
lo,
Is there a single central location for code and applications other than CVS? I'm talking about code that can't/wont be included in CVS for various reasons? Does the wiki have this sort of thing? I've done some code for the Cepstral TTS engine (bkw has done some updates too)
lo,
Is there a single central location for code and applications other than CVS? I'm
talking about code that can't/wont be included in CVS for various reasons? Does the
wiki have this sort of thing? I've done some code for the Cepstral TTS engine (bkw has
done some updates too) but apparently t
Hi,
From: "Peer Oliver schmidt" <[EMAIL PROTECTED]>
> >>I have 4569 opened and forwarded/NATed to my *. I am on the same network
> >>as the * server, a friend is remote. After about a minute you loose the
> >>connection.
>
> > This is another problem and it happens for me too (known bug, which
se
Dan wrote:
I have 2 DIAX phones behind two different NAT firewalls and the * box on
Cool. I am sure it has nothing to do with DIAX, but might be the
configuration on the * side. Using IAX(1) works fine, btw.
This is very interesting. It seems they still are a lot of bugs to be solved
in IAX2 ...:
An article I came across this morning..
http://www.itnews.com.au/storycontent.asp?ID=12&Art_ID=18128
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Debuuging SIP to a file:
asterisk -c | tee /tmp/sipdebug.log
then turn on 'sip debug' at the CLI
/O
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Hi,
There is any way to use the PSTN line callwaiting functionality (including
callwaiting callerID) with an X00P card?
When a second incoming call, on an internal ATA I hear the callwaiting tone,
but I don't know how to switch to the other caller through ATA->*->X100P.
More, the callerid is not d
hi.
I've gs working under NAT,
simply put nat=yes into sip.conf section if *,
then enable nat into the gs, without any stun server.
Matteo.
Il mar, 2004-02-03 alle 21:17, Tomas Prybil ha scritto:
> Hi all.
>
> Is it at all possible to have a GS B101 NATed with firmware 1.0.4.40?
> I've tried bot
> I had some similar problems with the X100P and our ATA-2. I also couldn't
> ever get the Nortel to recognize the DTMF, or get Asterisk to recognize
> DTMF
> coming through the Nortel. I wish I could say that I figured out a really
> cool way to make it work, but instead I moved on and interconn
> -Ursprüngliche Nachricht-
> I had to do a quick modification to chan_sip denying redirects to
> the (magic
> hardcoded) vm number.
>
can you tell me those changes ??
> Hope for a new release where you can either set the vm, vm-listen
> separately,
> or at least disable the redirect "fe
The problem is when replacing a Nortel system. The existing phones become
useless, so we're looking at either using totally IP based phones or using a
channel bank with different office phones.
The only problem is finding an IP phone that is decent for business,
supports multiple lines (at least 2
Hi everyone, I have
TE410P connected via E1 to telecom and another E1 to my internal Ericsson PBX.
All calls from PBX to telecom pass through Asterisk. Our telecom is providing us
with billing information during a call, and I would like to transfer this
information to PBX and be able to show
Hi,
> > I have 2 DIAX phones behind two different NAT firewalls and the * box on
one
> > of the phones network.
> > It works for me.
>
> Cool. I am sure it has nothing to do with DIAX, but might be the
> configuration on the * side. Using IAX(1) works fine, btw.
This is very interesting. It seems
Hi Dan,
iax2 to asterisk, and dial, but as soon as asterisk tries to do the
native
BTW: I have the same problem.
I have 2 DIAX phones behind two different NAT firewalls and the * box on one
of the phones network.
It works for me.
Cool. I am sure it has nothing to do with DIAX, but might be the
What are my support options for CALEA with Asterisk?
Ryan
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Hi,
- Original Message -
From: "Peer Oliver schmidt" <[EMAIL PROTECTED]>
> Greg,
>
> >>my Linux iptables firewall, on a private network. Both boxes cann
register
> >>iax2 to asterisk, and dial, but as soon as asterisk tries to do the
native
>
> > "a private network" -- as in a NATed netwo
Greg,
my Linux iptables firewall, on a private network. Both boxes cann register
iax2 to asterisk, and dial, but as soon as asterisk tries to do the native
"a private network" -- as in a NATed network? Maybe canreinvite=no or
nat=yes will do the magic you need.
I think he is using the IAX2 protoc
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