>b) If you are planning to use SIP make sure you configure it properly
to work with NAT.
>SIP has a lot of issues with NAT. The alternative is using IAX but the
IAX deskphones
>arent as feature rich as the SIP phones. Also take into account the
cost of deploying IP phones.
Intent wrote that he
Hi,
We are using asterisk version 1.0.5.
We have registered two UA's with asterisk.
(Registration was successful)
UA1 <---> * <> UA2
Now, UA1 subscribes for UA2 to asterisk.
asterisk sends NOTIFY to UA1 with UA2's state as open.
But if UA2 gets un-registered then,
asterisk is no
On Sun, 13 Mar 2005, C. Tomlinson wrote:
> How does recording work..i file per person, or are they all muxed into one,
> or can you specify?
I have not used it myself, but the docuemntation looks like one file for
the whole conference. Each member can be recorded with the "Monitor"
application
I thought this patch was added into the 1.04 and later source code?
On Mon, 14 Mar 2005 07:24:44 +0200, Dimitris Kounalakis
<[EMAIL PROTECTED]> wrote:
> I never managed to make outgoing calls to broadvoice without the
> following patch to the file channels/chan_sip.c
> it comes from http://edvina
Or open up a firewall rule to allow access to the IP that asterisk is on
from the otherwise isolated
subnet.
You could also make the asterisk box 'multi-homed', ie, put a 2nd nic in
it and plug it into
both subnets, then in sip.conf set the bindip to 0.0.0.0 (all interfaces).
In essence whateve
intent wrote:
I'm looking to use Asterisk to replace my current PBX system. I'm in
Australia so I need to use Austel approved equipment. My plan is roughly
as follows:
- Get a box with a suitable card and install Asterisk
- Connect our existing PSTN lines to the Asterisk box
- Get suitable softphon
Peter Bowyer wrote:
On Mon, 14 Mar 2005 00:27:12 -0500, Andres <[EMAIL PROTECTED]> wrote:
Deti Fliegl wrote:
Hi there,
all that started by investigating what happens if SIP clients are
calling anonymously.
The problem: Every client who is registered as a regular user with
username and secr
On Mon, 14 Mar 2005 00:27:12 -0500, Andres <[EMAIL PROTECTED]> wrote:
>
>
> Deti Fliegl wrote:
>
> > Hi there,
> >
> > all that started by investigating what happens if SIP clients are
> > calling anonymously.
> > The problem: Every client who is registered as a regular user with
> > username an
Hello,
I am new in linux and also suse i have a fxo card but its not working
the errors are:
Zaptel Configuration
==
Channel map:
Channel 01: FXS Kewlstart (Default) (Slaves: 01)
1 channels configured.
Notice: Configuration file is /etc/zaptel.conf
line 143: Unable to open
I'm looking to use Asterisk to replace my current PBX system. I'm in
Australia so I need to use Austel approved equipment. My plan is roughly
as follows:
- Get a box with a suitable card and install Asterisk
- Connect our existing PSTN lines to the Asterisk box
- Get suitable softphones/ip phones
Deti Fliegl wrote:
Hi there,
all that started by investigating what happens if SIP clients are
calling anonymously.
The problem: Every client who is registered as a regular user with
username and secret can fake any callerid in subsequent INVITEs.
Asterisk does not apply an accountcode or calle
I never managed to make outgoing calls to broadvoice without the
following patch to the file channels/chan_sip.c
it comes from http://edvina.net/broadvoice/ and it is the only fraction
that it is still needed for outgoing calls.
It does not cause any problems with other sip devices that are conne
Hello,
Thank you guys it works with
setting overlapdial=yes,immediate=no .
Regards
Nauman Bin AliPeter Svensson <[EMAIL PROTECTED]> wrote:
On Fri, 11 Mar 2005, Joe Antkowiak wrote:> > Hello > > > > Well i think that overlapdial=yes would be required if i am trying to dial> > from the asterisk si
Matthew Boehm wrote:
On the "no compatible codecs" error, do a "sip show peer 621" and see what
codecs it has listed.
vpbx*CLI>
* Name : 621
Secret :
MD5Secret:
Context : inhouse
Language :
AMA flags: Unknown
CallingPres : Presentation Allowed, Not Screened
>Of course I am not a kernel expert, so .. please be patient.
>I am investigating on my zaptel/zapata problem.
>As the main error message asterisk quits on mentions <'/dev/zap/channel':
No such file or directory> I went
>peeking over there.
>[Asterisk Verbose Error
>Mar 13 20:43:35 WARNING[5779]
You will probably get it Monday morning. Thats what happened to me when I
ordered on a Friday noght.
- Original Message -
From: "David Uzzell" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Sunday, March 13, 2005 8:10 PM
Subject: [Asterisk-User
Hello,
I know of someone that is thinking of spending $20,000 on a new
voicemail system because their vendor is end-of-lifing the system they
have now. I mentioned that maybe Asterisk could do what they need, at a
much lower cost. Reliability is, of course, critical -- which brings up
the topic
I'm actually talking to someone I have high respect for on this topic
right now, he has me convinced. I'm going to try it out.
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Thanks Rich... Your comments were very interesting. Now it's a bit clearer.
Thx!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson
Sent: Domingo, 13 de Marzo de 2005 08:07 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subjec
> > Guys. I have a few IAX2 connectivity questions that maybe somebody can
> > clarify to me:
> >
> > I have my * server and another one with a friend. We are both inside nat and
> > doing port forwarding:
> >
> > * -> nat -> internet -> nat -> *
> >
> > Now, what I dont understand is this, why
Does anyone know how long the orders take?
I ordered some a couple of days ago and it said normally 24hours, and I
am guessing that the weekend cause's some delays but it did not say
anything abouy that.
Any one got any ideas on how long generally over the weekend it takes?
Thanks
David
On the "no compatible codecs" error, do a "sip show peer 621" and see what
codecs it has listed.
For the changes: when you do a "make update" there should be new copies of
sample configs inside asterisk/configs/ that you can read through.
-Matthew
> From: Ronald Wiplinger <[EMAIL PROTECTED]>
> R
When do you use trunk then?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Rich Adamson
Sent: Domingo, 13 de Marzo de 2005 07:37 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] IAX2 and asterisk servers link
> Guys. I have a few IAX2 connectivity questions that maybe somebody can
> clarify to me:
>
> I have my * server and another one with a friend. We are both inside nat and
> doing port forwarding:
>
> * -> nat -> internet -> nat -> *
>
> Now, what I dont understand is this, why FWD needs to be co
Matthew Boehm wrote:
Are you sure that NAT is set correctly everywhere? I sometimes forget to set
the phone to be NAT aware.
That is weird that 'sip show peers/users' doesn't show the phone both times.
Have you stopped/started asterisk since these changes? Do it again just to
make sure.
The only th
Did you ever get arounnd this issue? I am seeing the same thing,
On Sun, 13 Mar 2005 00:04:54 +0545, Vicky Shrestha
<[EMAIL PROTECTED]> wrote:
> Thanks,
>
> I have that already in my /etc/hosts
>
> But it's still not working :(
>
> On Saturday 12 March 2005 03:48, Rich Adamson wrote:
> > For e
So far nobody has answered this post... Anybody has seen this error before?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robert
Goodyear
Sent: Domingo, 13 de Marzo de 2005 04:22 p.m.
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject:
Firefly?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Roman
Zhovtulya
Sent: Domingo, 13 de Marzo de 2005 04:49 p.m.
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: [Asterisk-Users] Looking for a free SIP/IAX softphone with IM
and
This shows you don't know how centericq works and how its configured :).
Its pretty secure if you knowwhat you are doing.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Strom Carlson
Sent: Domingo, 13 de Marzo de 2005 04:45 p.m.
To: Asterisk Users Mail
FWIW I get the same exact error at the end of every VM session as well,
thus:
-- Playing 'vm-intro' (language 'en')
-- Playing 'beep' (language 'en')
-- Recording the message
-- x=0, open writing:
/var/spool/asterisk/voicemail/default/501/INBOX/msg format: wav49,
0x8186370
Vtech and Uniden
http://www.voip-info.org/tiki-index.php?page=VOIP+Phones#id416800
Jim
James H. Thompson[EMAIL PROTECTED]
- Original Message -
From:
Chuck
To: asterisk-users@lists.digium.com
Sent: Sunday, March 13, 2005 1:28
PM
Subject: [Asterisk-Users]
cordle
C F wrote:
how are you telling the cisco what the password is? TFTP?
TFTP (SIPmacaddress.cnf)
you will not see anything on * CLI unelss you do sip debug
And after "sip debug" I saw (among other lines):
[...]
Retransmitting #5 (NAT):
SIP/2.0 407 Proxy Authentication Required
[...]
SIP/2.0 401 Unauth
I have a fairly current CVS build of asterisk running on SuSE 9.2. You
need to get rid of the stuff that gets installed with the system and
then install the zaptel stuff. Works fine for me, but I do get warnings
about unsupported modules and tainting of the kernel.
The wiki has an entry on SuSE
thanks
On Mar 13, 2005, at 3:42 PM, C F wrote:
Panasonic makes a system that has a very good 2.4 GHz system, with
multi cordless and multi site support that can be easily hooked into
asterisk using either FXS or FXO (CO ports on the Panasonic to FXS
ports on Asterisk looks the best). It's the TAW-8
thanks
On Mar 13, 2005, at 3:41 PM, Jason Becker wrote:
Chuck wrote:
does anyone know of a 2.4 or 5 ghz cordless phone system that has an
ip base station?
Uniden has the UIP1868:
http://www.uniden.com/productsupport2.cfm?product=UIP1868
But there's no documentation to speak of.
Regards,
--
Jason
Email to SMS, it goes both ways. I frequently email people from my
phone, and they can email me right back. With T-Mobile, the email
address is the phone number 1(areacode)[EMAIL PROTECTED]
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h
Peter,
How does recording work..i file per person, or are they all muxed into one,
or can you specify?
What do you mean by new primitives?
C
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Peter Svensson
Sent: 13 March 2005 21:56
To: Asterisk Users Mail
Chuck wrote:
does anyone know of a 2.4 or 5 ghz cordless phone system that has an ip
base station?
Uniden has the UIP1868:
http://www.uniden.com/productsupport2.cfm?product=UIP1868
But there's no documentation to speak of.
Regards,
--
Jason Becker
Director & CEO
Coalescent Systems Inc.
403.244.80
Panasonic makes a system that has a very good 2.4 GHz system, with
multi cordless and multi site support that can be easily hooked into
asterisk using either FXS or FXO (CO ports on the Panasonic to FXS
ports on Asterisk looks the best). It's the TAW-848 from panasonic
with each cell site taking up
does anyone know of a 2.4 or 5 ghz cordless phone system that has an
ip base station?
thanks
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Thanks,
Are you doing it by setting the lowest cost?
Is there anything in Asterisk which does it?
Thanks,
robert
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Wojciech Tryc
> Sent: Sunday, March 13, 2005 12:44 PM
> To: Asterisk Users Mailing L
Hello,
Could anyone recommend something similar in functionality and
user-friendliness to SJPhone, but that would additionaly have IM and
presence support?
Thanks a lot,
Roman Zhovtulya
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ht
On Sun, 13 Mar 2005 16:13:04 -0600, Anton Krall
<[EMAIL PROTECTED]> wrote:
> I already have this workling for remote linux admin. For example, each linux
> box has it MSN user and I have them on ly MSN list. So if I need to reset a
> server, I just send an IM via MSN to the user with the keyword re
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.
hich mailbox. I can then call back my * box and
listen to the messages, I like this better than the callback feature
b/c I can do it on my time.
> >Just take a look at this:
> > http://story.news.yahoo.com/news?tmpl=story&cid=569&ncid=738&e=1&u=/nm/20050313/tc_nm/col
Check out centericq on freshmeat. Lets your linux box be on MSN, ICQ, AIM,
etc.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jess Coburn
Sent: Domingo, 13 de Marzo de 2005 03:18 p.m.
To: Scheda; Asterisk Users Mailing List - Non-Commercial Discussion
S
I already have this workling for remote linux admin. For example, each linux
box has it MSN user and I have them on ly MSN list. So if I need to reset a
server, I just send an IM via MSN to the user with the keyword reboot and
the server runs the command.
If you need any help on setting this up, l
Guys. I have a few IAX2 connectivity questions that maybe somebody can
clarify to me:
I have my * server and another one with a friend. We are both inside nat and
doing port forwarding:
* -> nat -> internet -> nat -> *
Now, what I dont understand is this, why FWD needs to be configured in
iax.co
Have any of you tried this?
http://asteriskathome.sourceforge.net/
I'm thinking of using this version. I'm debating between it and
Knoppix with Asterisk thrown in there as well. I'm a linux newbie for
the most part, but can get around and get done what I need done with
help here and there, but I
On Sun, 13 Mar 2005 16:18:06 -0500, Jess Coburn <[EMAIL PROTECTED]> wrote:
> So you basically want an SMS or IM callback app right?
>
> One way to do this would be send an email to an address like
> ([EMAIL PROTECTED]) and have a cronjob query/pop this email
> address for your specific message and
On Sun, 13 Mar 2005, dean collins wrote:
> Taking yourself off mute is one of the more important requirements for
> broadcast conferences.
That is available already: enable the star-menu with the 's' option.
Entry 1 (the only one) allows the user to mute himself.
> I probably dial in to about 3
ave not seen the other way around.
>Just take a look at this:
> http://story.news.yahoo.com/news?tmpl=story&cid=569&ncid=738&e=1&u=/nm/20050313/tc_nm/column_pluggedin_dc
> Most providers have an SMS to email gateway. To send a message to any
> SprintPCS phone use: [EMA
On Sun, 13 Mar 2005, Matthew Asham wrote:
> > Whatever gave you that idea? Most operators have an interface allowing
> > reception of sms:es over internet. The protocols may be strange (they are)
> > and the pricing models vary greatly, but there are many receive interface
> > to sms:es.
>
> I
Taking yourself off mute is one of the more important requirements for
broadcast conferences.
I probably dial in to about 3 conference calls a week (using commercial
services) where the default is everyone in the call is on mute and then
you press star to talk - some automatically take you off or
> On Sun, 2005-03-13 at 11:14, Peter Svensson wrote:
>
>> Whatever gave you that idea? Most operators have an interface
>> allowing reception of sms:es over internet. The protocols may
>> be strange (they are) and the pricing models vary greatly, but
>> there are many receive interface to sms:es.
A co-worker installed the card and when the driver was loaded the lights
went red! (instead of just turning off) This is a big step forward,
however I won't be testing asterisk with the card until tomorrow.Fingers
crossed. :)
Cheers,
Peter
-Original Message-
From: Eric Bishop
On Sun, 13 Mar 2005, Jess Coburn wrote:
> So you basically want an SMS or IM callback app right?
>
> One way to do this would be send an email to an address like
> ([EMAIL PROTECTED]) and have a cronjob query/pop this email
> address for your specific message and then when it finds it have it
> cr
On Sun, 13 Mar 2005, C. Tomlinson wrote:
> I couldn't find, for example, a variable containing the current conference
> name.
>
> If I had those I agree it would be simple in the dialplan; just listen for a
> key eg 2, then when pressed kick user from conference, and immediately
> rejoin using a
So you basically want an SMS or IM callback app right?
One way to do this would be send an email to an address like
([EMAIL PROTECTED]) and have a cronjob query/pop this email
address for your specific message and then when it finds it have it
create a .call file to call you and connect you to wha
ws.yahoo.com/news?tmpl=story&cid=569&ncid=738&e=1&u=/nm/20050313/tc_nm/column_pluggedin_dc
Most providers have an SMS to email gateway. To send a message to any
SprintPCS phone use: [EMAIL PROTECTED],
for Verizon use: [EMAIL PROTECTED] I don't know for
the others. There is also
Try merging both and use type=friend
Julian.
On Sun, 13 Mar 2005 21:07:06 +0100, Pepe Aracil <[EMAIL PROTECTED]> wrote:
> I only can get outgoing or incoming calls work well, but not both.
> How can i solve this problem?
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-- Forwarded message --
From: C F <[EMAIL PROTECTED]>
Date: Sun, 13 Mar 2005 15:46:16 -0500
Subject: Re: [Asterisk-Users] Sipura 841 issues
To: Master Abi <[EMAIL PROTECTED]>
Well Cisco 7960 doesn't, Polycom IP300 (from which I conclude any
Polycom) doesn't, Sipura SPA-841 doesn't
On Sun, 13 Mar 2005 13:25:10 -0600, Anton Krall
<[EMAIL PROTECTED]> wrote:
> There is a program for linux called centericq, this program is for
> connecting a linux box to aim, icq, msn, etc something like trillian.
> Anyway, this centericq lets you define external commands that can run when
> you
Hello Stefan,
Thank you for response. it helped me to solve it.
The statement order was the problem here.
I checked the source and I found that chan_capi separes config for
different capi controllers with
the directive "devices". So the "devices" must be the last directive for
each controller.
Afte
Hello. I'm new in the list and sorry for my poor english :)
I have this two entrys in the sip.conf file, one for incoming calls (vtele_in)
an the other for the outgoing calls (vtele_out)
-- piece of sip.conf ---
; entry for incoming calls
[vtele_in]
type=user
context=sip-in
host=voztele.com
disa
Of course I am not a kernel expert, so .. please be patient.
I am investigating on my zaptel/zapata problem.
As the main error message asterisk quits on mentions <'/dev/zap/channel':
No such file or directory> I went peeking over there.
[Asterisk Verbose Error
Mar 13 20:43:35 WARNING[5779]: chan
On Sun, 2005-03-13 at 11:14, Peter Svensson wrote:
> Whatever gave you that idea? Most operators have an interface allowing
> reception of sms:es over internet. The protocols may be strange (they are)
> and the pricing models vary greatly, but there are many receive interface
> to sms:es.
I've
On Sun, 13 Mar 2005 13:32:41 -0600, Steven Critchfield
<[EMAIL PROTECTED]> wrote:
On Sun, 2005-03-13 at 11:15 -0600, James Taylor wrote:
Yes, the meetme can be part of it.
I was thinking more of a "classified ad" chat line, you know the
male-female thing:
"...If you are a man looking for a woma
tim panton wrote:
On 13 Mar 2005, at 11:21, Darrell Berry wrote:
hi:
Just starting out with *, and I'm planning to heed the advice to start
simple and small, but the goal i'm aiming for eventually is:
*-based pbx for 10-20 seat small business, based in the UK. Users will
have PoE SIP hardphones.
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
Hello *Martijn,
Thank you for your response.
*That was my opinion too, it looses the context due to a bug, and can anyone
confirm it also?
But I have no output from the command "Show channels", and it happens so
quickly that it is impossible to issue the command before falling to the default cont
Hi,
Sorry, I relaise I could run head but didn't want to move across yet.
I have found documentation on meetme fairly lacking; hence my n00bish
questions. I realize you can get people to enter the same conferences but
with different options; however I got stumped on a couple of things:
Wiki info
Why not chown to the user asterisk is running under? That way you
don't give write access to everybody. AMP does that.
Julian J. M.
On Sun, 13 Mar 2005 13:31:12 -0600, Matthew Boehm <[EMAIL PROTECTED]> wrote:
> As such, chan_zap is unable to work due to bad permissions. Is it safe to
> simply ch
As a security precaution, we run asterisk as non-root. When zaptel /dev/
devices are created, they get owned:groupd as root:root with rw-r--r--
permissions.
As such, chan_zap is unable to work due to bad permissions. Is it safe to
simply change permissions on all /dev/zap/* stuff to rw-rw-rw ? Is
At the risk of sounding like a closed source fan (I'm not) I do think
you should
at least consider Oracle for this job.
I built a system a few years ago which takes a constant stream of
entries from a number (<100)
of remote systems analizes them and generates reports
(see http://www.westpoint.l
There is a program for linux called centericq, this program is for
connecting a linux box to aim, icq, msn, etc something like trillian.
Anyway, this centericq lets you define external commands that can run when
you send it a message containing certain words. Also, you can define a
"system" call in
Using CVS-HEAD libpri, CVS-HEAD zaptel, CVS-STABLE asterisk.
Everything compiled fine. No problems loading chan_zap.so.
Incomming calls to PRI work fine. Outbound is a different story:
-- Executing Dial("SIP/64.72.107.4-4122fb40", "ZAP/R1d/18005551212|60")
in new stack
-- Called R1d/1800
I've been using asterisk MGCP for a good year now, and to do what you are
asking, would require
hacking the source. It would be nice to be able to edit the mgcp.conf to
send or not send specific parameters, but right now, it is not available.
But a better question is why? I have used several M
On Sun, 2005-03-13 at 11:15 -0600, James Taylor wrote:
> Yes, the meetme can be part of it.
>
> I was thinking more of a "classified ad" chat line, you know the
> male-female thing:
>
> "...If you are a man looking for a woman, press one..."
> "...If you are a woman looking for a man, press two
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 interface allowing
r
Hello,
On Sun, Mar 13, 2005 at 12:21:42PM +0200, Dimitris Kounalakis wrote:
> I am trying to configure asterisk 1.0.7pre to get incoming calls from an
> ISDN line using an AVM fritz PCI 2.0 with Chan_capi 0.3.5. My problem is
> that the context is not recognised in the /etc/asterisk/capi.conf
I
That is if you have a local connection into an SMS network.
I have heard this is available on some European ISDN systems.
In the US, good luck. Outside of getting a GSM phone and
connecting it to your system via a serial port and some sort
of GSM SMS application.
Your best bet is an IM system. I
James, it's a piece of cake, you should be able to do this in an
afternoon with about the same for the billing app.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of James
Taylor
Sent: Sunday, March 13, 2005 12:15 PM
To: Asterisk Users Mailing List - Non-C
On Sun, Mar 13, 2005 at 06:44:52PM +0200, Dimitris Kounalakis wrote:
> Thank you for your response Marco.
>
> I do. The problem is that all incomings calls from ISDN are handled by the
> default "s" extension in the context [default] and not by an "s" extension
> in the context [isdn] or by the ms
I found what that was, http://ruk.ca/article/1832 is the link. Not
exactly what I want, but I also found this.
http://www.voip-info.org/wiki-Asterisk+cmd+Sms
That seems to be what I want. I can send an SMS message, and then
configure it to call me once it recieves it.
On Sun, 13 Mar 2005 12:17:
Are you sure that NAT is set correctly everywhere? I sometimes forget to set
the phone to be NAT aware.
That is weird that 'sip show peers/users' doesn't show the phone both times.
Have you stopped/started asterisk since these changes? Do it again just to
make sure.
The only thing I can say is t
Grandstreams do, Sayson 480i does, so does all softphones. They should,
because how are you going to se what you typing.
Not having a backlit display is bad design.
C F wrote:
I haven't seen a sip phone that once connected will show the digits
pressed on the screen.
My SPA 841 doesn't give me any
Roy Sigurd Karlsbakk wrote:
Thanks, that's what I want to do.
Any chance of me getting my hands dirty with this code? Please?
Sorry, there's no way I can distribute it in the state it's in, it's got
bunches of other stuff in it that can't be easily separated out, and
most of it does not work.
___
Does ASTCC has functions like press a button and
topup another card before it runs out of credit and check the balance which
talking (by pressing a * 8 or some number) or if i make a mistake while entering
the pin press ## and re enter.
is there a place where i can find all the key pad
fun
Hi
I installed ASTCC and got it working, when i enter
the pin number and dialled the number needed, it says this call will cost point
20 cents per minute, can i get a message like you have 40 minutes and 30 seconds
than giving the per min rate ?
Thank You
Kani
__
I think I saw something a while back that would allow Asterisk to check AIM
to see if a user of an extension was in front of their desk or not then send
to VMail or whatever.
This may be a start for you but I can't recall the name of it or where the
info is.
--
Wholesale Private Label Internet
Does anyone know of a program/extention to asterisk that would allow
me to either text message my asterisk box or IM it from AIM on my cell
phone to allow it to call me? I've been looking with google yet can't
find anything. I don't code, so I'm SOL there, so I'm looking for
something premade. I pl
Robert,
Nufone, but it all depends on the destination.
For some is gafachi, for some is VoicePulse etc..
W
- Original Message -
From: "Robert Augustyn" <[EMAIL PROTECTED]>
To: "'Asterisk Users Mailing List - Non-Commercial Discussion'"
; "'Justin Richards'" <[EMAIL PROTECTED]>
Sent: Sunda
Hi, This app looks perfect for what I need. Are there any instructions how
to install?
- Original Message -
From: "Dan Austin" <[EMAIL PROTECTED]>
To: "Asterisk Users Mailing List - Non-Commercial Discussion"
Sent: Saturday, March 12, 2005 1:15 PM
Subject: [Asterisk-Users] ANNOUNCEME
You mean that if on a certain queue, your agents are using SIP or IAX
phones, and you want to do a check so that when a cllers tryies to
get into
the queue, if no agent is logged in, do something else with the caller
instead of hanging up?
Actually, I think he wants to go one step deeper, and if t
James Taylor wrote:
Just installed ASTCC, got it working.
I've noticed that only part of the sounds come from Allison.
Someone (male voice) has recorded the necessary balance,call cost, etc.
So, there's this mix of male/female announcements.
Is this new or am I missing some sound files?
It seems to
Wojtek,
What are you using for your primary route?
robert
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] On Behalf Of
> Wojciech Tryc
> Sent: Sunday, March 13, 2005 9:31 AM
> To: Justin Richards; Asterisk Users Mailing List -
> Non-Commercial Discussion
> Su
Just installed ASTCC, got it working.
I've noticed that only part of the sounds come from Allison.
Someone (male voice) has recorded the necessary balance,call cost, etc.
So, there's this mix of male/female announcements.
Is this new or am I missing some sound files?
--
James Taylor
MetroTel
3505 S
Setup an IVR to take care of the menu you described, and use different
meet-me rooms per destinations.
Marc
James Taylor wrote:
Yes, the meetme can be part of it.
I was thinking more of a "classified ad" chat line, you know the
male-female thing:
"...If you are a man looking for a woman, press
Dear all
I am looking for a per minute DID # in spain..either IAX/SIP
Thanks
Jer
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