Hi,
Is it to possible to route incomming call using dnis information to specific extension or section in the extensions.conf ??
like my users dial my asterisk box using different toll free numbers, but i can't offer them unique services.
Plz suggest that if it is possible ? if possible then
azher:
simple to do, assuming numbers are being passed through on dnis, in
the relevant context (from zapata) put
exten => 6601122,1,Hangup #users dialing here bye-bye
exten => 5551122,1,Playback(beep) #users here (pun intended) beep
- wasim
On Sun, 17 Aug 2003, Azher Amin
Yeah, I haven't really had time the last couple of weeks to follow up on this.
I'd be happy to have any of the patch in CVS if that were so desired, but
before heading down that road I'd be curious to know if anyone besides me has
had a successful time actually using the patches.
At the very le
Does anyone know what the Grandstream Budgetone is going for $$$ in the
US? I didn't immediately see pricing on the phones page.
AJ
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RRP: $75 for 101, $85 for 102
- Original Message -
From: <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Sunday, August 17, 2003 1:22 PM
Subject: [Asterisk-Users] Grandstream Budgetone
Does anyone know what the Grandstream Budgetone is going for $$$ in the
US? I didn't immediately s
Hi,
I have been using chan_oh323 with a latency issue even on the same network.
I am now trying chan_h323 and can only get one way audio. I am testing
using SJPhone -> SJPhone, and also SJPhone -> 7960 (SIP).
Any ideas? Must be something obvious that I am missing?
Thanks.
Regards,
Stev
Title: Message
Did something change
in lock.h lately? I get all kind of ast_mutex errors when trying to compile chan
capi 0.24c with the latest asterisk code
Betel ConsultancyAbelenlaan
19
T: +31 20 640 30181185 RT Amstelveen
E: [EMAIL PROTECTED]The
Netherlands
I haven't tried the patches, but they sounds very useful! My 2 cents...
BTW, there have been some recent bug fixes to Voicemail2, so you might
want to test them against recent CVS (8/16 or later)
On Sunday, August 17, 2003, at 11:16 AM, Brad Bergman wrote:
Yeah, I haven't really had time the l
ast_pthread_muxtex_* functions were changed to ast_mutex_* for the
possibility of debugging the mutexes with gdb.
regards
Martin
On Sun, 17 Aug 2003, Michiel Betel wrote:
> Did something change in lock.h lately? I get all kind of ast_mutex errors
> when trying to compile chan capi 0.24c wit
I have read loads of the emails on the subject but have not read a setup I
could use.
I have a windows box using Winproxy to dial up the internet through a DSL
modem. My internal network I am using while developing.
My internet network has been renumbered so that I could test using
Nikotel4Mac.
Hi,
We have an asterisk box, with 2 nics, one with internal addressing,
and the other with a public address. The firewall (iptables) is configured for
nat routing. Now we want to allow this box to receive sip registrations from the
internet. Does anyone know if you can use iptables to allow
$75 for the single ethernet port version and $85 for the dual ethernet
port version.
You can get two for $129 at www.sipphone.com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Sunday, August 17, 2003 8:23 AM
To: [EMAIL PROTECTE
I'm sorry I'm coming in on the rear end of this but is there a difference
between Voicemail and Voicemail2 in asterisk? If so what is it?
AJ
On Sun, 17 Aug 2003, Paul Cheng wrote:
> I haven't tried the patches, but they sounds very useful! My 2 cents...
>
> BTW, there have been some recent bug
I use asterisk behind my Linux firewall with no problem. For IAX I have
the firewall forward udp port 5036 to the asterisk box. Really simple nat
setup. My asterisk box also connects to other asterisk/IAX servers
through nat from behind the firewall. I'm not sure about sip at all but I
know
Hello All,
Been completely obsessed for the last two days with VoIP and Asterisk - running on 2
hours sleep and coffee - sorry if this is a little scattered...
Okay, I've got a small start-up company that installs traditional PBX (Nortel mainly)
systems, data network infrastructure, commercia
Hi all,
I'm looking for recommendations on ethernet switches for a new
install. Ideally would want switches with at least 24 ports,
ideally with a GE uplink, and that support PoE (power over
ethernet) on every port. I've seen lots of switches, and lots of
"power hubs", but the combination, whic
On Sunday 17 August 2003 12:17, [EMAIL PROTECTED] wrote:
> I'm sorry I'm coming in on the rear end of this but is there a
> difference between Voicemail and Voicemail2 in asterisk? If so what
> is it? AJ
All new features are going into Voicemail2. Voicemail (1) is being
EOLed.
Among the new fea
Are these locked to the service, though? Look what vonage managed .. :)
-d
At 12:36 PM 8/17/2003 -0400, you wrote:
$75 for the single ethernet port version and $85 for the dual ethernet
port version.
You can get two for $129 at www.sipphone.com
-Original Message-
From: [EMAIL PROTECTED
Hi Mike,
Cisco makes PoE switches, either at the Cat 29xx or
the Cat 35xx levels. The 29xx don't have gige uplinks, but
the 35xx's do via GBIC interfaces. Meaning you will also need
to get a GBIC media converter depending the media type (copper
fiber, etc)
And of course Cisco makes PoE based ph
They don't seem to be locked to the service. I've
ordered several sets and changed the configs to use
my AsT box with no problems
On Sun, Aug 17, 2003 at 12:49:26PM -0500, denon wrote:
> Are these locked to the service, though? Look what vonage managed .. :)
>
> -d
>
> At 12:36 PM 8/17/2003 -
On Sun, 2003-08-17 at 19:49, denon wrote:
> Are these locked to the service, though? Look what vonage managed .. :)
>
According to a thread on the ser list no they are not.
But when I checked on the site for a price I found the shipping would be
around 75US$ to France and then French customs wou
The Snom VoIP phones support PoE and Nortel makes switches:
http://www.nortelnetworks.com/products/02/bstk/switches/baystack_460/
I am not certain that they are compatible, as I have not used the Snom
phones and have only used the Nortel switches with PoE adapters at the
other end to power wireles
Not sure about making the ports dynamic but I just opened up inbound UDP ports
1-2 as in rtp.conf (these can no doubt be changes to what ever you needs
require) which has been working well so far..
> Hi,
>
> We have an asterisk box, with 2 nics, one with internal addressing, and the o
To follow up on this, the Cisco Switches and the Cisco phones will work
together to create two vlans: a prioritized vlan for the phone traffic,
and a secondary 10/100 link for a computer which can be attached to the
phone's second switched ethernet port. Some config is needed in the
switch and ro
Just for the record and to possibly help with others who get BudgeTone
phones.
My asterisk box is behind NAT, and I use Vonage, NuFone, and
iconnecthere for my "POTS backhaul."
On the front end I have an ATA186, a Digium TDM20, and now a BudgeTone 102.
The BudgeTone definitely has issues wrt t
Hiyas..
Have any of you tried the HP300 phone and got it to work with asterisk ?
(sip)
/Mike
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At 09:02 PM 8/16/2003 -0600, you wrote:
installed about a week ago. I'll go pull to see if there is more
recent code
I had this problem two weeks ago, and it seems to be gone as of yesterday.
However, I did end up changing two things at once: I used a newer CVS
version, and I changed the DTMF mo
Hi,
I'm planning to buy a new ISDN-PBX (I hope this is the right term for an
ISDN phone system). I would also like to connect it to asterisk. As far
as I know there is no ISDN card where I can connect an ISDN-Phone to
directly working together with asterisk (please correct me if I'm
wrong). So wha
What did you change the DTMF mode to? Where can I find documentation
with all the possible options in the config files?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ernest W.
Lessenger
Sent: Sunday, August 17, 2003 4:06 PM
To: [EMAIL PROTECTED]
Subject:
On Sun, Aug 17, 2003 at 03:44:21AM -0500, Gnophone Support wrote:
> Hello, and thank you for registering at gnophone.com. Your login
> information is listed below:
>
> Username: miernik
> Password: ***
> IAX Phone Number: 17002916107
>
> Please login as soon as
Why not get IP phones and use Asterisk as the PBX from the start??
Will probably save you a LOT of money and many headaches...
> Hi,
>
> I'm planning to buy a new ISDN-PBX (I hope this is the right term for an
> ISDN phone system). I would also like to connect it to asterisk. As far
> as I kno
At 04:14 PM 8/17/2003 -0400, you wrote:
What did you change the DTMF mode to? Where can I find documentation
with all the possible options in the config files?
Check out the demo config files - I think those are the only "complete"
documentation in existence. It looks like I actually removed the l
I have removed all the dmtfmode= statements from my sip.conf to begin
with. Earlier today I downloaded and compiled the latest CVS and from my
testing right now it seems to work a lot better.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ernest W.
Lessen
How do I get the asterisk compile to produce the cdr_mysql.so module,
assuming this is what I need to get CDR's into my mysql database.
Is "load => cdr_mysql.so" what I put in the modules.conf file as it says?
It looks like asterisk loads the modules in /usr/lib/asterisk/modules, but
there is no c
On Sunday 17 August 2003 17:25, Scott Stingel wrote:
> How do I get the asterisk compile to produce the cdr_mysql.so module,
> assuming this is what I need to get CDR's into my mysql database.
You need to install the mysql client libraries and headers. Follow the
appropriate instructions for your
An 'ISDN' phone? You mean a handset that actually support
ISDN? Didn't know they had these, and if they do I'm sure
they wouldn't be cheap.
Are you talking BRI or PRI? I'm guessing BRI which means
you're right, there is no 'card' to go in an Asterisk box
that will do this.
However, you mig
I'm not near my * box at the moment, so can't check this,
but IAXTEL isn't down again, is it? Can you ping
iaxtel.com.
J
On Sun, 17 Aug 2003 22:12:30 +0200
Miernik <[EMAIL PROTECTED]> wrote:
*This message was transferred with a trial version of
CommuniGate(tm) Pro*
On Sun, Aug 17, 2003 at 03
Bugga, it's definately a monday. Replied to the wrong
subject. (see below).
J
On Mon, 18 Aug 2003 09:42:05 +1000
"Jamie Carl" <[EMAIL PROTECTED]> wrote:
*This message was transferred with a trial version of
CommuniGate(tm) Pro*
An 'ISDN' phone? You mean a handset that actually
support ISDN
Does anyone have any recommendations for a cordless phone that uses SIP
(or IAX)? It doesn't have to use 802.11b, but that would be appreciated.
Thanks,
Nathan
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Quoting Oliver Brandt:
> Hi,
>
> I'm planning to buy a new ISDN-PBX (I hope this is the right term for an
> ISDN phone system). I would also like to connect it to asterisk. As far
> as I know there is no ISDN card where I can connect an ISDN-Phone to
> directly working together with asterisk (pleas
Hello,
Just installed Asterisk on my Redhat 8 box. With
all the many configuration options I can see, my question is:
where do I start ?
Is there any documentation I can work with apart
from the draft pdf I downloaded ? or is this sufficient ?
Kind regards
Dayo
On Sun, 2003-08-17 at 17:55, Nathan wrote:
> Does anyone have any recommendations for a cordless phone that uses SIP
> (or IAX)? It doesn't have to use 802.11b, but that would be appreciated.
I think you're only solution is going to be the Cisco ATA-186, an
analog-to-SIP device. Or, you could use
Well, there is the Netlink phone line from Spectralink:
http://www.spectralink.com/products/nl-wts.html
My wireless bridge supported prioritizing traffic from spectralink phones,
and they apparently will do SIP (and are 802.11b phones), but those are
the only cordless phones I know of that will do
Just for the record and to possibly help with others who get BudgeTone phones.
My asterisk box is behind NAT, and I use Vonage, NuFone, and
iconnecthere for my "POTS backhaul."
On the front end I have an ATA186, a Digium TDM20, and now a BudgeTone 102.
The BudgeTone definitely has issues wrt th
At 5:18 PM + 8/17/03, [EMAIL PROTECTED] wrote:
Hello All,
Been completely obsessed for the last two days with VoIP and
Asterisk - running on 2 hours sleep and coffee - sorry if this is a
little scattered...
Okay, I've got a small start-up company that installs traditional
PBX (Nortel main
Is there not support for H.323 endpoints in asterisk? The Symbol
NetVision phone comes to mind, but it does not support SIP.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nathan
Sent: Sunday, August 17, 2003 7:55 PM
To: [EMAIL PROTECTED]
Subject: [Asteri
Hi,
Did anyone have any comments on the below problem - or did you (shong
ching) manage to solve this? I have the same issue - any assistance would
be great. Thanks.
Regards,
Steven Thomas
I too tried to get festival 1.4.3 working, but no luck. It appeared to compile just
fine. The server started okay. I don't have a sound card on the server so I can't
test it without *. When I invoke the * festival application, the asterisk process
gets 100% CPU time with no sound. I need to
I just came back from Daytona. My niece wanted to go to the reptile show.
I ended up watching a water-do boat race with my nephew.
Let's hook up before I go on vacation on the 29.
Uriel
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Andrew
Joakimsen
Sent: S
My setup
dial 100 -> phone -> Asterisk -> iax -> asterisk -> x100p
if I dial 100 I get a dial tone on the far x100
I need to be able to flash the x100p card over the internet, when I press
flash, then dial *0 and flash the x100p
On Sat, 2 Aug 2003, Martin Pycko wrote:
> Well when you use you
i also encountered this problem
i'm not too sure either but i don't think codec has to do anything with it
for i tried mix and matching but to no avail.
so for the meantime, try adjusting the tos for oh323 and i think you could
live with it
by the way, are you running cvs?
- Original Messa
not sure what you mean by 'are you running cvs'?
What does the TOS setting do?
Regards,
Steven Thomas
"Kelvin Chua"
set
ipTos=lowdelay
in oh323.conf
and try to see what happens. (of course this would mean your switch should
have the ability to detect TOS bits in the packet headers)
what version of * are you using? did you check against cvs?
- Original Message -
From: "Steven Thomas" <[EMAIL PROTECTE
[apologies for no line wrap; config lines at bottom]
I have mentioned on several threads here that the Monitor application doesn't do
exactly what one would expect: the originating and answering legs of a call are
unsynchronized by the duration of the interval that it takes for the answering leg
Heya,
I'm just playing with a SIP phone. When I log into my queue from a
SIP agent it appends some sort of data to the end of the SIP name.
e.g. SIP/sablaptop-2ac0. I didn't add the '2ac0', asterisk did. When I
log out of the queue, it uses a different ID (e.g. SIP/sablaptop-5207)
and therefore d
Hi,
A cordless phone with support for both PSTN and IP will be available at the
beginning of 2004.
See the link:
http://www.eutecticsinc.com/products/consumer.html#IPP700
BR,
Dan
P.S. In this moment I have an ATA186 with two DECT cordless phones which
works like a charm with Asterisk.
- Or
On Mon, 2003-08-18 at 08:24, Sebastian Filzek wrote:
> Heya,
>
> I'm just playing with a SIP phone. When I log into my queue from a
> SIP agent it appends some sort of data to the end of the SIP name.
> e.g. SIP/sablaptop-2ac0. I didn't add the '2ac0', asterisk did. When I
> log out of the queue,
Title: Message
Does anyone know of
a Java based SIP client and if so have has anyone used it.
I found JAIN at https://sip-communicator.dev.java.net/ but
have not tried it yet.
Rgds,
Stuart
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