am working on it for a client, and yes as Steven said, I think the meetme
will do itnow just to figure out the billing part :-)
On this note as anyone thought of premium line SIP addresses...I know
this may sounds strange, and SIP--SIP is normally free (this I feel
will change once voice
Rich Adamson wrote:
Looks like a couple of problems here. I don't believe the Cisco phone
handles md5, so remove that line.
As I told before, tried 3 different approaches:
1) password; md5;
2) password, no md5;
3) no password, no md5.
Only the third one worked. Trying to give SOME security, I
Checkout
http://www.alwaysonvpn.com/
Umar.
On Sun, 13 Mar 2005 11:21:02 +, Darrell Berry [EMAIL PROTECTED] 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
If asterisk is going to be modified to support LiveVoip expectations,
then yet another Dial option would need to be implemented to
force ringback to occur as an audio stream for iax only. Guess
one could open a bug report for both LiveVoip and Asterisk, but
not likely to be addressed
On Sat, Mar 12, 2005 at 03:26:40PM -0500, William Suffill wrote:
NuFone service bills in industry standard billing increments, which
are: six (6) seconds for the US48, sixty (60) seconds to Mexico and
fifteen (15) seconds to the remainder of the world.
From: http://www.nufone.net/tac.html
I can see errors on the console, g.729 and ilbc works no problem.
I endup moving VoIPjet to the secondary route.
Wojtek
- Original Message -
From: Justin Richards [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Saturday,
OK here is a
possible tricky one.
I have a rocom door
entry systemwhich connects to an FXS port on my TDM400P card. When
the door button is pressed it initiates an 's' extension which dials a number of
SIP extensions. When a SIP phone is picked up the user can speak to the
person at the
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.
Make sure you use the 'g' flag in the Dial command to go on in the context
after a hangup. Now whether the tone will
The g with your dialplan of playtones should realy work. Also try the
h extension.
I've seen a few door opener modules, but I haven't seen one where you
have to press a key to hangup. Look thru the docs of the door oepner,
maybe it can be disabled.
On Sun, 13 Mar 2005 10:10:04 -0500, Andrew
Well, it is still working today; I had tried so many sip.conf
combinations that it seems like it had to be something outside of
sip.conf.
If DNS is reliable, it does not seem like the hosts entry would be
needed anyway.
JDC
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
how are you telling the cisco what the password is? TFTP?
you will not see anything on * CLI unelss you do sip debug, this will
realy give you much more info (I guess in your case you will get a 403
forbbiden).
On Sun, 13 Mar 2005 22:38:46 +0900, Hermann Wecke [EMAIL PROTECTED] wrote:
Rich
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 backlit on the display. So I think that not.
On Sun, 13 Mar 2005 23:31:03 +1100, Master Abi [EMAIL PROTECTED] wrote:
Hi
Just 2 issues I have with SPA841.
1. I
Hi,
Thanks. I have already tried various options in from the wiki, but they
don't work in my situation.
I do not think the announce option works as I am using STABLE, not
HEAD...huess I have to wait for it to make it into stable.
I am creating dynamic conferences, using the 'd' option.
The m
Hi Guys.
I have some questions regarding how to interconect * server with each other.
We are 3 asterisk servers and we added each other as friend on iax.conf.
So far everything was working or appeared to be working fine until:
1. One of the server changes it IAX2 port each time it reboots, why
thanks to everyone offering sussgestions and pointers in answer to my
questions about SME VoiP/* in the UK; i'll update you on my progress as
and when this ends up as an implemented * solution (and i totally take
on board the recommendation to look at 2nd hand CCME -- i should have
said at the
I just downloaded the new astcc and it includes now a new field in the
list of the cards: Brand
Great!
How can I use it in the dialplan?
bye
Ronald
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Someone once said YOU CAN'T BE TO RICH OT HAVE TOO MUCH BANDWITH
1 How much do you have? How many phone calls and how many other users
on your connection?
2 Go to http://testmyvoip.com/ and test your bandwith
Jeff
Date: Sun, 13 Mar 2005 09:23:35 +0100
From: Edward Banfa [EMAIL PROTECTED]
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 send
On Sun, 13 Mar 2005, C. Tomlinson wrote:
Thanks. I have already tried various options in from the wiki, but they
don't work in my situation.
I do not think the announce option works as I am using STABLE, not
HEAD...huess I have to wait for it to make it into stable.
Or you can run cvs
Matthew Boehm wrote:
You may not have most recent CVS. You should have this in your sip.conf:
You are right, ... but the sip.conf will not be updated anyway, if I do
not want to loose all my settings.
rtcachefriends=yes
; Cache realtime friends by adding them to the internal list
; just like
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 msm numbers as extensions in the context
[isdn].
So, what is the reason for the
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...
...If you are not sure, press three...
...If you don't care, press
Dear all
I am looking for a per minute DID # in spain..either IAX/SIP
Thanks
Jer
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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
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
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
Subject:
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
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
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
asterisk-users@lists.digium.com
Sent: Saturday, March 12, 2005 1:15 PM
Subject:
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'
asterisk-users@lists.digium.com; 'Justin Richards' [EMAIL
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
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
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
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
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.
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
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
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
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 msm
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 -
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.
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
Is
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
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...
...If
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
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
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
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
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 ?
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
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
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
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
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
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 woman,
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
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]:
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
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.
After
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 send
-- 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.
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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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. Just take a look at this:
http://story.news.yahoo.com/news?tmpl=storycid=569ncid=738e=1u=/nm/20050313/tc_nm
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
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 mute
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
create
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
quote who=Matthew Asham
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
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, 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've been
not seen the other way around.
Just take a look at this:
http://story.news.yahoo.com/news?tmpl=storycid=569ncid=738e=1u=/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
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
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 then
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
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.conf
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,
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
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=storycid=569ncid=738e=1u=/nm/20050313/tc_nm/column_pluggedin_dc
Most providers have an SMS to email gateway. To send a message to any
Robert Hajime Lanning wrote:
quote who=C F
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.
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 reboot
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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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 List -
does anyone know of a 2.4 or 5 ghz cordless phone system that has an
ip base station?
thanks
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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
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.
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
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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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
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
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
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
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]
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
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
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
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:
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 everyone
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
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
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]
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