codecs.
I did work on converting G.729 to G.711 to disk storage in real time and
that took about 3% of a Xeon CPU for full duplex.
Memory wise each convert call might have used 640KB in buffers and
trash, but not much.
Race Vanderdecken
Code Tyrant
-Original Message-
From: [EMAIL
I have done per user context programming for asterisk in the recent
past.
I am a professional contract programmer but my rates are reasonable and
you will get code that works.
Please contact me if you would like my help on this.
-Race
Race Vanderdecken
Code Tyrant, Inc.
[EMAIL PROTECTED]
828
help
programming. I could use some advice on telephone circuits.
Race Vanderdecken
Code Tyrant, Inc.
Somewhere near Asheville, NC.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Gary H.
Thompson
Sent: Sunday, February 18, 2007 2:05 PM
To: asterisk
.)
Race Vanderdecken
Code Tyrant
Somewhere near Asheville, NC.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Douglas
Garstang
Sent: Friday, October 06, 2006 1:49 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [asterisk-users
The O'reilly Book Asterisk, the Future of Telephony is a good book.
I don't operate Asterisk; I just mess with the internal code and stuff
like that. This book helps me configure and understand how asterisk
works from the user side.
Race Vanderdecken
Code Tyrant
[EMAIL PROTECTED]
828 221 2636
makes up 1 second or x seconds...
Later.
Race Vanderdecken
Code Tyrant
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tobias
Wolf
Sent: Friday, September 22, 2006 5:05 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk
then 32,000 bytes would fall under the 2 second
threshold.
How you get the size of the file is up to you.
What language are you using? API?
-Race
Race Vanderdecken
Code Tyrant
Somewhere near Asheville, NC.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
the orphaned .txt file bug.
-Race
Race Vanderdecken
Code Tyrant
[EMAIL PROTECTED]
828 221 2636 vonage
828 699 2361 cel
Somewhere near Asheville, NC.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Arnd
Vehling
Sent: Saturday, August 26, 2006 4:13 PM
Hmmm,
Let me see.
Skype has 54 million registered users that the online retailer
eBay can add to its marketing arsenal. The initial payment of $2.6
billion values those registered users at just more than $48 each
Skype has current revenue of about $60,000,000 a year.
Because the server does not want dead or unconnected phones that might
move.
If the phone does not send a REGISTER every so often, periodically, then
the server will assume the phone is no longer available to send calls
to.
Unlike the Government, Asterisk will not send checks to dead people.
Just a guess,
101 to 111 could be seen as dialing prefixes.
Check your extension file for stuff like that. Maybe it thinks you are
trying to dial overseas?
Race
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Jeff
Brownlee
Sent: Monday, August 22, 2005
Nope, I have the same problem, nothing.
I jumped on the ISP for not being able to get my mail. Ooops.
Race
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Huddleston, Robert
Sent: Monday, August 01, 2005 2:47 PM
To: 'Asterisk Users Mailing List -
Nope, I have the same problem, nothing.
I jumped on the ISP for not being able to get my mail. Ooops.
Race
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Huddleston, Robert
Sent: Monday, August 01, 2005 2:47 PM
To: 'Asterisk Users Mailing List -
www.footnotess7.com is now open to begin the creation of SS7 that can be
used with asterisk.
Sign up and list which and what parts you would like to work on.
Race V.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael D
Schelin
Sent: Friday, July
www.footnotess7.com is now open to begin the creation of SS7 that can be
used with asterisk.
Sign up and list which and what parts you would like to work on.
Race V.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mike M
Sent: Thursday, June 09, 2005
www.footnotess7.com is now open to begin the creation of SS7 that can be
used with asterisk.
Sign up and list which and what parts you would like to work on.
Race V.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven
Critchfield
Sent: Sunday,
Adam makes some good points.
What I found while doing this was that users would learn to only forward
emails, or give out the TTS email address to people who knew that the
email was going to be read TTS.
Adding IM text to the TTS on asterisk would be good.
Don't worry so much about the
Yes there is a way to do it with SIP, but I don't think it is in
Asterisk.
I have the code archived somewhere if you would like to see it.
As an aside to the general audience;
If I put code snippet's like this on my site, to do one easy thing, what
would you be willing to pay? $5, $10
Race
I worked with Intellivoice. They did VAD, voice activated dialing, on
the switch. You had to dial a number, speed dial on the cel, to get the
reco. It worked with any phone. Their research should that speech
recognition was more accurate then DTMF dialing.
They were doing voice pattern
Ed,
Please let me know how you make out. I am sort of keeping tract of what
asterisk needs for speech.
I am not working a project yet, just trying to get a feel for what
people need before I start making new stuff.
Race the tyrant Vanderdecken
Race at code tyrant dot com
-Original
If you need to modify the C source code for voicemail to add 0 or other
numbers I can do that for you.
Race the tyrant Vanderdecken
Race at code tyrant dot com
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Carlos
Alperin
Sent: Friday, July 08, 2005
Your customers are not going to like this.
You have to change the way you bill for calls.
For $1 your customer gets 60 seconds worth of phone time. However you
have to also charge, like the Bells used to, for setup and teardown
time. Remember the operator used to say Deposit $1.85 for the
It gets even better here in the US.
You can prepay for you cell service, but your unused mintues expire
after 90 days, and you forfeit your balance.
Greed is good.
Race
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Tony Hoyle
Sent: Thursday, June 16,
Title: Message
Greetings,
You have stumbled on to
one of the most troublesome flag for newbies;
autocreatepeer.
http://www.voip-info.org/tiki-index.php?page=Asterisk+sip+autocreatepeer
in your sip.conf file add a
line in the [general] section autocreatepeer=no
Now people can
Also check out the CISCO GKTMP API, that is their gatekeeper api. There
might be some cool stuff you might like to know.
Race the tyrant Vanderdecken
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Simone
Sent: Sunday, June 12, 2005 2:13 PM
To: Asterisk
Also subscribe to the asterisk-dev mail list. Watch it for a couple of
days before you ask a question or they will eat your lunch.
Pick a single thing you want to change in the PBX, and then learn how to
code for that. Something really simple like adding a parameter to a conf
file is a good place
Are they running on a windows server? :)=)
Maybe it has the Monday Morning Blues. (I can't get it to talk either.)
Race
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Damon
Estep
Sent: Monday, June 13, 2005 11:05 AM
To: asterisk-users@lists.digium.com
Yeah, if you get the Microsoft Partners Newsletter emails they reported
the 75 GB expansion today.
Increased Storage Limit in Exchange Server Standard Edition
Get more out of mission-critical email. In the fall of 2005 the storage
limit for Exchange Server 2003 Standard Edition will increase
the voicemail system to enable it to
utilise other entities as the store, e.g. a pop3 server or an imap
server rather than just flat files on disk (which should remain an
option).
That way it doesn't matter where they listen to them or delete them
from?
Steve
From: [EMAIL PROTECTED] on behalf of Race
Curious, which codec are you using with Sphinx?
The smaller the bandwidth, generally, the harder it is to do
recognition.
Sphinx4 is a tool built on JAVA (one moment while I clear my throat and
spit to get the coffee taste out of my mouth.)
Write me offline; I am curious about doing batch reco
Good things are happening.
Another aside from having done this before:
If configuration requires the user to do anything or the user
to load a piece of software it won't work.
Everything must be configured from an admin consol or it won't
work. You will go crazy trying to keep
Good Idea, but not practical as it breaks the second commandment of IT
user management.
1. Thou shall not require any brain cells on the part of the end-user.
2. Thou shall not require any settings to be set on the users equipment.
...
More rules to follow...
Race the tyrant Vanderdecken
Aye, there's the rub.
Now having said that, obviously we can't delete the message from the
local store of the POP3 client after it has been already downloaded, but
we are not talking about that, are we?
1. Thou shall not require any brain cells on the part of the end-user.
2. Thou shall not
(a) Has anyone cracked this nut (or started on it)?
Been there, done that.
No, really. I was the architect on the Premiere Tech, no PTECH,
Orchestrate system. www.orchestrate.com (pardon the plug, but I am not
endorsing one way or the other.)
I also did the MAPI work for Persona, but that was
IMAP vs. Exchange
I would be very weary of using IMAP against an Exchange Server. I have
not touch it for years but IMAP and Exchange did not play together
really well back then.
Has anyone actually used real IMAP to talk to an Exchange server from a
Linux client lately?
If your Linux client
Give it a break you freakin
Cry Baby
Race the Tyrant Vanderdecken
-Original Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Karl J. Vesterling
Sent: Tuesday, May 31, 2005 8:05
PM
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Subject: RE:
Hmmm,
You are going to price yourself out of the market if you go with
hot swap.
If I understand you correctly that is.
Your residential gateway sits in a home and connects to the
internet to do VoIP calls for the owner.
What is your cost for this gateway? Doing
Hmmm, it seems to me that a Mr. James Hendrix once showed how a chicken
could be cooked on an electric guitar during a public concert.
Race ;)
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ian
Pattison
Sent: Tuesday, May 24, 2005 4:23 PM
To:
Yes, I have done what is called a Zombie list. I save the current
registrations list every time a new registration comes in.
When asterisk recycles I send a SIP message to everyone in the zombie
list asking them to reregister with asterisk. Part of the SIP protocol.
Mind you this only works with
A Rube Goldberg type solution is to send a text message to the cell
phone, reply to the message if you want the call to forward to your cell
phone.
You do have to keep the ZAP connection waiting for the SMS/text round
trip.
Race they Tyrant Vanderdecken
-Original Message-
From: [EMAIL
Could you set up an NFS directory that is shared between the servers?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David Shaw
Sent: Sunday, May 22, 2005 9:50 AM
To: Asterisk Users Mailing List
Subject: [Asterisk-Users] 2 Asterisk boxes sharing dial
You have to put entries in sip.conf
Race the Tyrant Vanderdecken
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michele
O-Zone Pinassi
Sent: Friday, May 13, 2005 6:47 AM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] Asterisk newbie
I've
Curious,
How did you do the forward? Was it a script or programming in C?
Any output from debug?
Race The Tyrant Vanderdecken
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton
Krall
Sent: Saturday, April 30, 2005 8:00 PM
To: 'Asterisk Users Mailing
Sounds like a good idea to me. I would watch it.
Race Vanderdecken
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt Roth
Sent: Friday, April 29, 2005 2:42 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users
Thank you,
I have been watching with interest your postings.
While I have not read everything, I have stored your messages.
I think your contributions will inspire a new VoIP soft switch movement.
Race The Tyrant Vanderdecken
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL
Oh, Ronald, What have you done?
Fore sooth. Now we, the trembling masses of the not so great unwashed,
unknowing and must apologize to the great and powerful Asterisk Ether
list watchers of the Mighty Dave, all hail Dave, to be able to receive
an answer to our frustrations.
And,
I can give you some help with the SIP stuff.
Try it again with sip debug turned on and send the output back here.
It would be good to see the SIP messages that are being transferred.
The Asterisk SIP Stack is good, but not great. You might need to just
add or delete an option in the sip.conf
Yes, something similar has been done before with PSTN phones.
Callers would call a number, listen to a bunch of ads, then get X
minutes of free phone time for that one call.
This was about 5-10 years ago and was covered in news stories. I think
they might still be doing it.
Poking around on
Very Curious,
As a developer and Big Idea person I would like to know more about this.
I am kinda curious about the singing part.
Is this karaoke via asterisk?
Either way, if you send back a list of requirements it would help
decide how to do this in code.
Race The Off-key Tyrant Vanderdecken
Wow! What a great fight!
Let me egg you guys on.
Furthermore, (if you knew your history) MS had been doing funny
things with DOS / and windows to make it difficult for other windowing
systems and DOS clones to work with MS-DOS / Windows, further cementing
their market dominance.
As someone who
I guess we are not thinking about the global extent of asterisk.
$200 in a third world would be great money. You can almost buy a Dell
computer for that much.
But this is more like a $200 bounty to design, build and replace your
Yugo engine with a Ferrari engine. And I only get the money if and
Yes, if I understand what you are asking.
1. The Card User calls to your asterisk PBX.
2. Asterisk answers the call on line 1.
3. Asterisk places an outgoing call on line 2 bridging the lines.
(That is how it works in the SIP world.)
So you would need an FXO/FSO pair of lines to let them make a
Hmmm,
Hoot and Holler, hoot-n-holler, ARD: Automatic Ring Down, Hot
line and Private Line Automated Ringdown (PLAR)
You should think about VoIP via Asterisk.
Here is a quick search result on it
http://lists.digium.com/pipermail/asterisk-users/2003-March/008936.html
But not much
It looks to me, and I have wrong before, that what you are looking at is
more for scientific type computing clusters. Trying to simulate weather
or other complicated things.
And the price is not cheap.
(I am working on a design for a cluster trie, yes trie, not tree,
approach to doing
Just from long term experience it might be a heat problem.
Check the really basic stuff first.
The air flow might not be adequate for the box. Make sure your ribbon
cables and such are not blocking flow.
Two cards might draw too much power, causing the power supply to
overheat causing
Yes,
As a guy who has monkey with SIP and asterisk's chan_sip.c there
is a good example and starting point there.
It is a little messy, but it is a good teaching tool.
Write if you get stuck,
Race the Tyrant Vanderdecken
-Original Message-
From: [EMAIL
Yes it could be done.
You
just have to have Velveeta support the asterisk callback functions.
Copy
some of the functions in chan_sip.c so asterisk can
know how to load the Velveeta stack.
Then connect/tie/stitch in the pbx.c
callbacks.
Not
hard but not simple.
Much
of
As you are a new Linux and asterisk user you best path is to use a Linux
Distribution that is easy to install and setup.
I have heard that Mandrake is very good, but for me I like Fedora 2/3
from Red hat.
You will need an OS that has clear documentation in the form of books
and a well supported
This might seem really dumb but tack enough silence onto the back of
your file to make it five minutes long. Then the message play for 5
minutes and repeats.
Race The Tyrant Vanderdecken
This was a dumb idea.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf
Check sourceforge.net
I killed myself workin on Oh323 and pwlib and then foud the home site
was out of date.
Race The Tyrant Vanderdecken
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Guillermo
Salas M
Sent: Thursday, April 07, 2005 2:27 PM
To:
Okay, let me stir the pot with a hornet's nest,
Yes, you could use it with Asterisk.
You have to set asterisk up to look like the Vonage switch.
You have to spoof the switch.
You have to look at the traffic between the Vonage box and the
WAN.
But
Hmmm,
My question would be for testing Asterisk, can you run multiple
Asterisks?
If I have two Ethernet cards or interfaces setup, I should be able to
tell it to run in different directories by changing the modules.conf.
No?
Has anyone done this before?
Race the tyrant Vanderdecken
Also you can have several numbers terminating to one desk. That way you
can be the sales guy and the repair guy at the same time. I like the
Cisco 7960 because it has 6 lines.
Also it makes it easier to test, you only need one phone to make test
calls.
Race Vanderdecken
Tyranny Works
penalty is not that great compared
to the feature set of using sockets.
If speed of the calls cannot be maintained then it will switch
to a non-socket approach, but the code will hide all that in the
classes.
Race Vanderdecken
Tyranny Works
-Original Message-
From
-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Riddell
Sent: Tuesday, March 29, 2005 10:13 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Looking for SS7 design input
Race Vanderdecken wrote:
I am looking for input on what
Greetings All,
I am looking for input on what an SS7 interface to Asterisk
should look like and what it will need to be of any use.
If you don't want to help then don't whine and complain about
how you don't need SS7. All comments made in jest are welcome; points
will be awarded
Yes, this is true for SIP.
5060 is the port SIP normally listens for calls on. It is configurable
to another port.
The 10,000 - 20,000 range is configurable.
SIP will negotiate the ports.
Ports lower then 1024 are reserved for system use and often blocked
across open networks.
-Original
Yes, Ironic isn't it. That SIP/VOIP is so digitally clear that you don't
hear the pops, cracks and whistles of the old analog phones. The only
analog is from the human to the machine. The old analog phone humans
hear it, soon there will another generation of humans who have never
used an analog
Arrgh,
Why should I answer a Newbie question, they are thick!
Why is it so difficult to just ignore any question with Newbie in it?
Everyone has to start somewhere. At least the newbie found the list.
The worse you can do is kick sand in their face. No newbie's means no
new customers or
PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Steven
Critchfield
Sent: Wednesday, March 02, 2005 11:28 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: RE: [Asterisk-Users] Why should I answer a Newbie
question,therethick!
On Wed, 2005-03-02 at 11:01 -0500, Race
Okay,
About the secret, comment out the line. You do have to set the secret in
the phone. So when the INVITE is exchanged Asterisk will ask the phone
for the secret, no secret, no connection.
I don't have a polycom phone so that is about all I can help with.
Oh yeah, you need a context
Hmmm,
Looking directly at the .../channels/chan_sip.c code does not get any
clues.
Switch( resp )
...
...
case 480: /* Temporarily Unavailable */
case 404: /* Not Found */
case 410: /* Gone */
case 400: /* Bad Request */
case 500: /* Server error */
case 503:
The Cheapest way is to purchase 2
licenses, or in multiples of 2 If you need more, from Digium.
You will be beating a dead horse and a
dead carriage and a dead driver if you try to get around G729 licensing. You only
need a license for each answer and originate session that uses g.729
I am developing voicemail and SIP and RAIDUS code for Asterisk Code on
the Fedora Core 3 and having no problems.
I am running on an Intel Pentium 3, 1.5 GHz, mother board stuck inside
an old E-machine case and it is very happy... (I only wish I could find
a Okidata B4250 printer driver or a PCL-6
Greetings,
First, you are hereby admonished for asking programming/development
questions on the user list.
C does not have a way to do this directly. Yes, you could use some
preprocessing macro but the code would be a nightmare.
You actually gave yourself the answer in your questions.
You can
Hmmm,
Are you trying to collect digits during a playback that is not set to
listen for a digit?
From the coding side I know that depending on how the prompt is called
you can enter a digit and interrupt the prompt. Otherwise the prompt
will finish and then see the digits.
What you want is
Look at ast_exists_extension in pbx.c
I have tinkered a bit around them, but not directly coded to it. In the
app_voicemail.c code I think it is used to make sure the extension is
valid. There might be code in there to see if the registration is
active.
Look further I see that chan_sip.c has a
My two proposals are:
1. You can contact webvoip.com they are billing guys who do what you
need.
3. You can wait for IBS to become integrated into Asterisk.
Your question is a little vague as to what you need.
The more chatty among us readers will surprise you with help if you can
give more
Yes 7 lines on the SNOM 200 SIP phone.
Use a web browser to connect to your phone's IP address. There is a
world of things it can do via its built-in web server. Just don't change
the setting that says where to get the photos from, leave it as from
the phone.
Each line can be configured to
Please let me know the answer to this one.
I set up FWD today and I am having the same problem.
Thanks for the iax debug tip.
Race Vanderdecken
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Robert
Webb
Sent: Monday, February 21, 2005 9:32 PM
Tell me about.
I was on a project once that tried to use DELL PDA's with a soft phone
in them to be wi-fi back to asterisk. Unless it sat in a cradle,
attached to a wall outlet, you couldn't make it through a 10 minute
call.
Lucky if we got the PDA thing to stay awake for a call to come in.
One
Ouch,
Do you know how to use gdb, the Gnu Debugger?
That will give you a clue as to where the segmentation fault is coming
from.
Good, then let me move on to the insults and ranting.
1. Why are you running on Slackware?
Are you trying to prove a point or just enjoy being frustrated?
20, 2005 1:26 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Segmentation fault {Writer given
gnu-lashing}
Race Vanderdecken wrote:
Good, then let me move on to the insults and ranting.
1. Why are you running on Slackware?
Are you trying
-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Race
Vanderdecken
Sent: Sábado, 19 de Febrero de 2005 07:15 p.m.
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] No Sounds
Correct.
-Original Message-
From: [EMAIL PROTECTED]
[mailto
- Non-Commercial Discussion'
Subject: Re: [Asterisk-Users] Segmentation fault {Writer given
gnu-lashing}
On February 20, 2005 01:11 pm, Race Vanderdecken wrote:
1. Why are you running on Slackware?
Are you trying to prove a point or just enjoy being frustrated?
Open Source is like Broad
chan_alsa.so failed!
[EMAIL PROTECTED] root]# Warning, flexibel rate not heavily tested!
Ouch ... error while writing audio data: : Broken pipe
And asterisk quits...
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Race
Vanderdecken
Sent: Domingo, 20 de
and causing
problems. Took the sound blaster out and eth1 is solid as a rock.
Race Vanderdecken
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Anton
Krall
Sent: Sunday, February 20, 2005 5:00 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion
If you are new to VoIP then by all means get phones that can be
controlled via a web page from the phone.
I will say that SNOM has done a great job with their web interface to
the phones. I would praise others, but all I have seen are the Sipura
and the Cisco ATA, both not real intuitive web
Hmmm,
Let's use the single step approach here.
I saw only a few of the prior posts for this problem, so please bear
with me as we start from square one. Also I do not know how the server
is being used and if you can do all these things.
Remember that the hair loss and dental fracturing
Okay,
A couple of things could be happening so let's run through a list. Your
questions are a little vague so I shall make my answer also vague.
1. Codec.
Are you allowing for and does the phone support the codec that
the sounds are in? (I.e. do you have a G.729 license for your
Good work Edward.
Sometimes it is not you but the machine. Probably a device driver that
was not kosher.
Race
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Edward
Banfa
Sent: Saturday, February 19, 2005 1:00 PM
To: Asterisk Users Mailing List -
Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Race
Vanderdecken
Sent: Sábado, 19 de Febrero de 2005 05:43 p.m.
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] No Sounds
Okay,
A couple of things could be happening so let's
. So if I understand you
correctly,
Ill do the sip debug but maybe trying to force both to use ilbc or
ulaw/alaw
might help so I can listen to the prompts right?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Race
Vanderdecken
Sent: Sábado, 19 de Febrero
have my
sip.conf with only one number configured as Race Vanderdecken suggested:
[8003]
type=friend
host=dynamic ;- This is supposed to allow registration, isnt it?
username=ariel
disallow=all
allow=ulaw
And only one extension in extensions.conf
exten
Look at FireFly and Kphone.
Asterisk with IAX to FireFly.
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Kiran
Vahaja
Sent: Friday, February 18, 2005 8:22 PM
To: asterisk-users@lists.digium.com
Subject: [Asterisk-Users] VoIP Test Samples to test
Okay,
Rule number one, Only One Variable allowed per setup.
Take out the secret= on the sip.conf
[8003]
type=friend
host=dynamic ;- This is supposed to allow registration, isnt it?
username=ariel
disallow=all
allow=ulaw
In fact, shorten the entire
I have a design that works for Follow-Me and Find-Me is anyone is
interested.
I can help you with the code, but don't ask me to check-it in to the
CVS.
Race The Tyrant Vanderdecken
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Nitesh
Divecha
Sent:
Hmmm,
I faintly recall having a similar problem a couple of months ago.
I think my solution was to assign the two strings to two separate
variables and then compare the variables.
exten = s,3,GotoIf($[${CALLERIDNAME:0:4} = PRV:]?5)
ABC=($[${CALLERIDNAME:0:4}
XYZ=PRV:
exten =
Greetings David,
PerlBox would not be usable for the level of service that is needed by
Asterisk to be viable Speech.
PerlBox is a vocabulary based recognizer, or I as I call it a grunter,
where you grunt something and it then does something cute.
Grunters depend on you creating a vocabulary
Greetings Mr. Weber,
Remember the rule in mathematics that is much easier to solve for one
variable.
You stateed you are having a problem with the 1088 extension. If look
like you are trying to make a call from the 404 extension to the 1088
extension.
1.
If you have 6 ATA's running shut 5 of
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