.
Then you will use less bandwidth and have a better sound upstream.
Marcel
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul
Fielding
Sent: woensdag 29 juni 2005 1:24
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users
I tried a Calgary DID with Link2Voip, but they never did get it working
correctly. My primary complaint with their customer service is that it was
basically non-existant. It took 2 weeks before a service guy even
responded to my problem, we fired a few emails back and forth, and then I
So I'm using a WRT54GP2-NA when I travel, as I
travel alot, to give me a phone at my hotel rooms, etc. During the
day or late at night the thing works great - best ATA I've ever
used.
However, in the mid-evening (when many business
travellers are at the hotel room doing work), the outgoing
bandwidth than the downstream..
-Greg
On Tue, 2005-06-28 at 13:00 -0600, Paul Fielding wrote:
So I'm using a WRT54GP2-NA when I travel, as I travel alot, to give me
a phone at my hotel rooms, etc. During the day or late at night the
thing works great - best ATA I've ever used.
However
I've found the quality to be consistently good - I've tried several other
carriers and LiveVOIP is the best I've tried. Their support is indeed a
bit blunt, but they get the job done well and quite timely.
Paul
- Original Message -
From: JD Austin [EMAIL PROTECTED]
To: Asterisk
I think it's a server/connection issue with the LiveVoip server. I'm
connected to their Winnipeg server and I get pretty much perfect calling,
all the time. A buddy of mine recently got setup on the Vancouver server
and is also experiencing choppy audio. He's in the process of asking if he
- Original Message -
From: [EMAIL PROTECTED]
The fact of the matter is that LiveVoIP has no customer service. They
don't care about small users or asterisk users.
The few times I've had issues, I've sent off an email and gotten a response
within 2 hours. This includes 3am on a Saturday
- Original Message -
The Grandstream HandyTone 488 has an FXO port.
I've never used it though.
I could be wrong, but I seem to remember reading up on the HandyTone and
deciding that it doesn't really act like a true FXO, as in calls come in and
go straight to Asterisk like an FXO, and
@lists.digium.com
Sent: Friday, April 22, 2005 4:35 AM
Subject: Re: [Asterisk-Users] livevoip callerid
I don't think it's correct to put dashes in the CIDNum.
MARK.
Paul Fielding wrote:
Hmmm... I still can't get name, though number works. Perhaps I'm missing
something?
context livevoip in iax.conf
- Original Message -
From: snacktime [EMAIL PROTECTED]
At $200 someone might be
willing to do the work if they know it's going to be open source, but
if it's a work for hire, $200 is extremely paltry.
I'm with you on that one. $200 might be an acceptable bounty to give
someone a bit of
Ok, Here's my ztdummy question. Forgive my ignorance. Everything I read
about ztdummy and zaptel cards describes them as being required 'for
timing'. But what exactly does this imply?
Eg. I have two separate boxes where I did the following:
- installed Linux (debian Woody)
- compiled a 2.4
I've actually had sales question oriented calls answered at 2am on a sunday
morning, 45 minutes I sent the email (I wasn't expecting a response until
monday). Technical email responses at wierd hours as well. No complaints
here. I'd send the email again in case it got lost in the shuffle.
Hmmm... I still can't get name, though number works. Perhaps I'm missing
something?
context livevoip in iax.conf that hooks me to livevoip
dial 9 in front of long distance number to dial livevoip instead of regular
LD.
snip
LIVEVOIP=IAX2/username:[EMAIL PROTECTED]
snip
exten =
clients use the same?
For inside clients it should be a charm!
Very nice job Paul, intercity dialing and everything well connected...
That
was a good story.. Thx for sharing.
Anton
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Paul
Fielding
Sent: Martes
I've actually used xten lite on a mac using GSM codec on a dialup
connection. Worked like a hot damn. I was quite surprised, actually.
Paul
- Original Message -
From: cmisip [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
- Original Message -
From: Anton Krall [EMAIL PROTECTED]
would like to hear some actual setups and how people are solving the nat
issue within scenarios like:
Asterisk - nat (ports forwarded) - internet - nat - multiple voup phones
I've been playing with this with my friends for awhile
I use Asterisk because I want the flexibility.
My mom uses Skype because it just works. Hey, my Mom can configure Skype.
I'll give $100 to the first person that creates a SIP client that my Mom can
configure.
Forget the fact that Skye's audio quality easily surpases any SIP client
I've ever
(On top of which, they charged me a $40 termination fee to terminate
my account - just a parting shot I guess).
People need to read the fine print more. From Vonage's website:
If you cancel after the first 14 days of service, you will be subject to
the $39.99 termination fee. If you return the
Make this another vote for Zap and IAX2 monitoring :)
Paul
- Original Message -
From: Thorben Jensen [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
asterisk-users@lists.digium.com
Sent: Thursday, March 17, 2005 12:10 AM
Subject: SV: SV:
I think first we would need to clarify the use of the term FXO in this
context. I'm not a telco expert by any stretch, but it appears to me the
term is used misleadingly sometimes. It seems to me that an FXO port is a
port that you can plug an external phone line into - it can then allow you
Anyone tried running Asterisk in production on a
vmware box? I'm considering the possibility and looking for
sucesses/failures...
regards,
Paul
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
Don't sweat it - it just so happens that you came into the fray just moments
after this list has had a big long drawn out argument about newbie etiquete
(sp?). You've just managed to get caught in the middle. Don't let it be
indicative of how everyone feels, and don't let it scare you away...
- Original Message -
From: Jay Milk [EMAIL PROTECTED]
You won't be able to send caller-id NAME with any PSTN termination.
That's just not how that works. Each CLEC looks up the name in some
mystical database based on the phone number. How to get that DB, I
don't know, but it sure would
Just pick up the handset and the speakerphone will
turn off automatically.
If the handset isn't hung up already, just hang it
up and pick it up again. Hanging up the handset won't hang up a
speakerphone call...
Paul
- Original Message -
From:
dean
collins
To:
*shrug*. Mine's been working flawlessly since I've had it (~month). The
only 2 issues I have are the ringback problem, and I can only send callerid
number info to them, not name info Guess we'll see how long it
lasts
regards,
Paul
- Original Message -
From: Tim [EMAIL
Ummm... This isn't a Digium Run list. This is a Digium sponsored list. My
understanding (someone please correct me if I'm wrong) is that this list *is
not* a Digium support list. This list is a forum for Asterisk discussion by
users. As such, I would suggest that all topics of discussion
- Original Message -
There is a LOT of traffic on this list about products that are not
supplied by Digium. Do you want to exclude those also?
The Sangoma guys typically handle support for their own product, even on
this
list. Atacomm's card hasn't hit the market yet. The Sipura people
- Original Message -
From: Andrew Kohlsmith [EMAIL PROTECTED]
I think where the problem comes in is that people take this forum to be
asterisk-biz half the time. I need X done **RIGHT NOW**!! I DEMAND
HELP!!
-- take it to -biz, there are dozens if not hundreds of consultants who
will
- Original Message -
From: David Brodbeck [EMAIL PROTECTED]
Sure. So say, I tried a Googling for X, but I didn't have any luck.
Then
I looked at pages X and Y in the Wiki, but couldn't find anything that
related to my problem. People are a lot more sympathetic if you
demonstrate
- Original Message -
From: David Brodbeck [EMAIL PROTECTED]
Well, sometimes that works. But I've been on a lot of lists where newbies
who thought they were being ignored started flaming people for not
responding to them, writing posts badmouthing the project, hijacking other
threads,
Ok, time for me to ask my own newbie question. :) I've done some digging
on ringback, and if I'm understanding it correctly, it's the ring tone that
the caller hears when dialing another person.
What exactly is it that people are finding now working with LiveVoip?
Everyone says 'ringback
- Original Message -
From: Ryan Laginski [EMAIL PROTECTED]
Anyways, I haven't found anyone that offers a toll free number that
works in Canada for 1.29 cents a minute. If there is others, please
let me know.
You're LiveVoip toll free number costs 1.29 c/min from Canada? My toll free
*.
I'm interested in what the technicals are here
regards,
Paul
- Original Message -
From: Robert Webb [EMAIL PROTECTED]
To: Paul Fielding [EMAIL PROTECTED]; Asterisk Users Mailing List -
Non-Commercial Discussion asterisk-users@lists.digium.com
Sent: Friday, March 04, 2005 12:06 PM
What I'm not trying to understand is how Ringback works in this context.
err, I mean what I'm now trying to understand.
Paul
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
http://lists.digium.com/mailman/listinfo/asterisk-users
To
- Original Message -
Look, don't answer lame questions if you don't want to. Flaming a newb
for being a newb is just mean. (they will eventually RTFM or STFW or
they will fail). This is the way of the open source community.
Here Here, I'm with you. I find it a constant source of
- Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Monday, February 28, 2005 2:39 AM
Subject: Re: [Asterisk-Users] How does the g.729 registration program work?
Paul Fielding wrote:
I could be mistaken, but doesn't the license tie itself to the nics on
the server? I believe
- Original Message -
From: Kristian Kielhofner [EMAIL PROTECTED]
Paul Fielding wrote:
I thought that it was 3 different times with different nics, but with the
same nic didn't count.*shrug*. No matter - if we can just copy the
license keys that's much easier... :)
Yes
You misunderstand. Ofcourse I need to run the register program on the
machine itself. The point is I build them from images and every now and
then I roll out a new image. My question is, what do I need to preserve
from the previous image to keep the licences. Obviously reformatting
the disk and
- Original Message -
From: James Bean [EMAIL PROTECTED]
I am going to now sit in a corner and go quietly insane while playing
the banyo with no strings.
I'd probably go insane, too, if I was trying to figure out how the heck to
play a banyo
;)
I'm running a TDM-400P with 2 x FXS and 2 x
FXO. I'm finding that there seems to be an odd relationship to
sound quality on the card to my local when connecting via a SIP
client.
When I'm on my local network, if I connect to
Asterisk via a SIP client (such as x-pro), and dial an outside
Are there any other relatively low cost analog
cards available? I'm interested in finding something that might work a bit
more reliably than the TDM-400P
regards,
Paul
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
trying to find something that works more reliably than this
card has turned out to be.
Paul
Cheers,
Jon.
On Sunday 13 February 2005 02:55 pm, Paul Fielding wrote:
Are there any other relatively low cost analog cards available? I'm
interested in finding something that might work a bit more
Pure guess... you're probably bumping into some of the same issues
that many of us TDM users are hitting. Seems like either an interrupt
handling (latency) or pci bus issue. You'll find hundreds of postings
relative to this over the last six months or so. Not everyone has
problems with the TDM,
I've seen the same behavior on my TDM400P. I solved it by simply scripting
a stop/start of the zaptel drivers and asterisk in the middle of the night
each night.
Of course, that might not be practical in a more seriously production
environment
Paul
- Original Message -
From:
We have deployed many (20+) IAXy's in the field. At a couple of
locations, the IAXy's have just stopped working after 1 or 2 days use.
No lights go on, no DHCP lease is renewed as far as we can tell, and of
course no dialtone and no registration with the server.
I bought two of them, both of them
- Original Message -
I see the sip user is an external ip. I would take a look at your QoS
settings on your router. Make sure the voice traffic is getting the
priority it deserves. Also, check for packet loss.
I'd still be wondering if there's something else. I, too, experience choppy
Shouldn't you contact your vendor for support and not a different
vendors support channel?
Um, I didn't think this was a Digium support channel. I thought this was an
Asterisk Users channel. Seems to me the question should be fair game.
(Sorry I don't have an answer to your question, though,
Low bandwidth
Low CPU utilization
Best audio quality
I think you might want to clarify that Best audio quality is in relation to
other highly compressed codecs. Certainly my (albeit limited) experience is
that g711 is much more clear than g729. Compared against gsm, for example,
however, the
I've seen lots of stuff go around about Grandstream
firmware levels (in my case specifically the BT101/102). I'm just
wondering what the currently accepted 'best' firmware version is to use?
After seeing stuff going around about buggy firmware I want to know what I'm
getting into before
- Original Message -
From: Sean Kennedy [EMAIL PROTECTED]
Likely, you are running into packet queue problems. As I recall, the
vonage device goes on the line before anything else, so it can shape the
stream to put it's bits first, ensuring it's packets get out in a timely
matter ( #1
I've actually given up on my two IAXy devices - one of them keeps loosing
it's connection and needs to be rebooted, the otherone keeps its connection
but periodically loses it's ability to send clear audio and needs to be
rebooted. I'm going back to the Grandstream ATAs that work just dandily
- Original Message -
From: Brian Capouch [EMAIL PROTECTED]
Paul Fielding wrote:
I agree I probably am having some packet queue problems, however i don't
think it's my only problem. My Vonage ATA adapter is actually further
behind the line than my Asterisk server. My configuration
Ok, I've seen discussion before on doing transfers
(attended and unattended), there seems to be much confusion over
it.
As things sit, I've been trying (unsuccessfully) to
do transfers with a zap channel (analog phone attached to TDM400). I have
no idea what I'm missing. My current
When I experimented with DISA, I found it to be very unreliable - sometimes
it would ignore my key presses and just keep giving dialtone, sometimes it
would work. I couldn't find a rhyme or reason to it. I ended up just
giving up and going with the silence
Paul
- Original Message
for conference
calling?
I know I need to play with three way calling here
also.
Lyle
- Original Message -
From:
Paul
Fielding
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Monday, January 17, 2005 6:12
PM
Subject
List -
Non-Commercial Discussion
Sent: Monday, January 17, 2005 8:20
PM
Subject: Re: [Asterisk-Users] transfers
with zap channel
Have you looked at features.conf?
Lyle
- Original Message -
From:
Paul
Fielding
To: Asterisk Users Mailing List
So far in my playing with Asterisk I've messed with
soft phones (x-ten, sjphone), hard phones (Grandstream 102), and ATA adapters
(Grandstream 286, Digium IAXy).
I've also got a Vonage line, using a Linksys
ATA.
None of the devices I've connected to my Asterisk
server have been able to
Hahawell the MWI is the blinking blue LCD. The message button
is reserved for future use Hang in there. There will soon to be some
upgrades and rumor has it that the conferencing feature will soon be
introduced so that conference button on the phone will soon be
working.
The message
[EMAIL PROTECTED]
It occurs to me, do you have the numbered extension set the same as the
context name for the phone in sip.conf? For example, in my sip.conf, the
context names for each phone are [7001], [7002] etc. However, this doesn't
necessarily need to be true. If it's not true, try:
that to
justify
the extra $100 or so you going to pay for a wifi sip phone?
Paul Fielding ([EMAIL PROTECTED]) wrote:
I think some people are missing the point. You can't throw your
cordless
phone in your pocket, go to your office, hotel or buddie's house, turn it
on
and get a signal. You can
- Original Message -
From: Eric Wieling aka ManxPower [EMAIL PROTECTED]
the mailbox= has NOTHING to do with extensions.conf at all.
[EMAIL PROTECTED]
voicemail.conf:
[happypeople]
666 = 1234,Happy Dude
sip.conf
[blah}
...
[EMAIL PROTECTED]
Boy, I had a blonde moment back there, I was
I think some people are missing the point. You can't throw your cordless
phone in your pocket, go to your office, hotel or buddie's house, turn it on
and get a signal. You can with a WiFi phone, however
- Original Message -
From: Kim Lux [EMAIL PROTECTED]
To: Asterisk Users
you need to set 'mailbox=extention' in the sip phone's context in
sip.conf
Paul
- Original Message -
From: Ronald Wiplinger [EMAIL PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Wednesday, January 12, 2005 11:13 PM
PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
asterisk-users@lists.digium.com
Sent: Thursday, December 30, 2004 6:14 AM
Subject: Re: [Asterisk-Users] IAXy reliability issues
On Thu, 30 Dec 2004, Gary wrote:
On Thu, 30 Dec 2004 00:12:51 -0700, Paul Fielding wrote:
I've
I still am not sure how to check and/or upgrade firmware.
So far the only way I've found to upgrade the firmware is to update the
Asterisk code to more recent code, which includes newer firmware for the
IAXy. The next time the IAXy connect to Asterisk, it will automatically
install the new
I've just picked up a pair of IAXy devices. They work fine except that they
keep going offline. As in, I plug it in, it connects to Asterisk, I can
dial and phone and all is dandy. Then, maybe 12h later, maybe 24, maybe 36,
maybe 48, I'll either try to phone the device and not get through or
I've been messing with some hardware sip devices and with softphones
(X-Lite, X-Pro and SjPhone). Compared to the hardware devices, the
softphones blow chunks (tm) for sound quality. the softphones are quieter,
crackly, and overal more difficult to understand the voice, while a sip
device
I'm interested in this, too. I find that when I use Xten or SjPhone
software locally the quality is quite good, but when I use it remotely
across the internet, I get quite a crackly response.
*however*, if I use some SIP hardware, such as a Grandstream 236 or an IP
phone (still use alaw just
07 December 2004 05:39 am, Paul Fielding wrote:
I'm in the middle of getting g729 to work on my server and running into
odd
stuff. The issue revolves around what appears to be a much talked about
(but not seeming to be much solved) issue of selecting which codec gets
used at a given time.
I have
and ulaw. if I allow
only those two codecs, the problem I have is what I mentioned previously
(see below), and I can't select which codec I want to use. The goal is to
pick the codec depending on whether I'm in a high bandwidth or low bandwidth
environment
regards,
Paul
From: Paul Fielding
I
X-Pro does not allow you to only enable one codec?
Ostensibly so. I can disable the codecs at any time in X-Pro. Problem is,
it doesn't seem to work. I can disable a codec, and then when Asterisk
connects, the codec will magically light back up and get used, even though
I've disabled it.
I'm in the middle of getting g729 to work on my server and running into odd
stuff. The issue revolves around what appears to be a much talked about
(but not seeming to be much solved) issue of selecting which codec gets used
at a given time.
I have two g729 licenses. I'd like to be able to
I installed x-lite on a new system, and can't make it work. it connects to
the asterisk server fine, but it's outgoing audio is messed. Incoming audio
is fine and dandy, but outgoing sounds like someone ran it through a voice
deepening machine that makes it so low and slow that it's
Anyone tried using a pocket pc with sjphone or x-ten over a cell phone
connection? I'd like to be able to connect using my cell phone data
connection, but so far I've come across bandwidth constraints - The closest
to success I've found so far is to use the GSM codec, but even then the
Whatever. I find it frankly more annoying to have people bottom post. I
use Outlook Express for my mail (as do millions of others), and the way OE
formats it's mail lends itself to top posting.When you bottom post, I
need to scroll way down the message to see your response, while when you
: [Asterisk-Users] Re: Top posting
Paul Fielding [EMAIL PROTECTED] lazily top-posted:
Whatever. I find it frankly more annoying to have people bottom post. I
use Outlook Express for my mail (as do millions of others), and the way
OE
formats it's mail lends itself to top posting.
As you seem
I've currently got Asterisk configured to take
incoming calls and send them directly to my voicemail. I'd prefer to keep
this approach rather than sending people to a menu first.
What I want to be able to do is have voicemail come
up, but if someone presses a key, such as 9 or 8 or perhaps
I want to authenticate to the phone system, then be
able to call an extension or dial an outside line. My preferred
method would be to use DISA, because a) it's non-verbal - ie. it doesn't talk,
just provides dialtone, and b) it provides dialtone.
However, it seems to be unreliable. when I
Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Friday, November 12, 2004 6:34 PM
Subject: Re: [Asterisk-Users] pressing a key to get out of voicemail?
Paul Fielding wrote:
I've currently got Asterisk configured to take incoming calls and send
them directly to my
I've currently configured incoming calls to
simultaneously ring an analogphone (via TDM400P) and two SIP
phones. I'd like to have it also simultaneously dial out the TDM400P
on a PSTN to ring my cell phone, and have the first one to answer win the
battle.
In my digging I've figured out
PROTECTED]
To: Asterisk Users Mailing List - Non-Commercial Discussion
[EMAIL PROTECTED]
Sent: Friday, November 12, 2004 6:56 PM
Subject: Re: [Asterisk-Users] Calling an outside number along side
otherinternal extensions?
Paul Fielding wrote:
I've currently configured incoming calls to simultaneously
I've managed to get my Asterisk server up and
working, and my TDM400p seems to be working fine, inside and outside
lines. I was pouring through the syslog looking for a different
issue when I noticed the following 'Prosilc Failed' message (9 lines
in):
Nov 11 17:34:36 natasha kernel:
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