i m using several IAX SIP softphones Now I've got an IP phone(Netphone) that supports H.323 protocol plz tell me how should i configure it to work with asterisk i m comfortable with sip.conf iax.conf but what should i do to use h323.conf??? do i ned to install something or should i
On Thu, 2005-06-23 at 14:41 -0500, Brian West wrote:
chan_woomera is another alternative h323 implementation.
visit www.pbxfreeware.org for more information.
Without being rude, why do we need another one? ie, why did you decide
that another one needed to be written, what are the advantages
Hello anyone who can help
I have two Asterisk boxes with identical hardware (Dev Production). I
recently rebuilt the DEV box using Fedora Core 3 and the latest CVS Head.
The hardware is an Intel CA810e, onboard everything with a PIII processor.
The config is pure VOIP using IAX2 ilBC with
Hi All,
I think by now everybody knows that LiveVoip went down, bankrupt etc
So please stop nagging about it and move on to some topics that really
matter.
If you want to discuss LiveVoip, get all together in a restaurant, eat,
drink and nag and wine about it as much as you want. But do it
On Mon, June 27, 2005 0:15, harry gaillac said:
I agree you.
Does asterisk (Digium) project provide a good
documentation ?
Does Asterisk Handbook has been released ?
When developpers improve Asterisk where are you
looking for help, mailing list, wiki, asteriskdocs,
...-:(
It's the job
Is it possible that a RTP session between two end users (so i
want to use asterisk as a signaling proxy and bypass RTP sessions)?
I used canreinvite=yes but it didnt work.
Description from asterisk conf. File;
(canreinvite=yes
; allow RTP voice traffic to bypass
In article [EMAIL PROTECTED],
Rob Thomas [EMAIL PROTECTED] wrote:
In my experience, several seconds of delay becomes apparent over time
when
using an internal clock source. Seems its a clocking/timer issue.
Yes. Meetme can have horrible issues with timing. This _has_ been fixed.
If you
In article [EMAIL PROTECTED],
Tzafrir Cohen [EMAIL PROTECTED] wrote:
On Mon, Jun 27, 2005 at 11:48:00AM +1000, Rob Thomas wrote:
In my experience, several seconds of delay becomes apparent over time
when
using an internal clock source. Seems its a clocking/timer issue.
Yes. Meetme
In article [EMAIL PROTECTED],
Brian Capouch [EMAIL PROTECTED] wrote:
Tzafrir Cohen wrote:
This is documented at http://www.aussievoip.com.au/wiki-AMP-Zaptel
What's wrong with the standard 2.6 ztdummy?
How does HEAD zaptel interact with 1.0 asterisk?
While we're at it, and
Marcel van Kaam
I agree with you Francesco.
It is as simple as this: If you want everything for free then sit on the
corner of the street with your hands open (called begging).
Open source means you also have to do something for it. You don't want to do
something for it, then you pay and
Hi,
it helps to have configured and working kernel sources installed.
Configure your kernel sources for the running kernel and then run
make in the kernel source dir to build the necessary scripts.
You dont have to wait until the kernel is compiled.
best regards
Klaus
--
Klaus-Peter Junghanns
Hi,
that is a bug in libpri. You will sometimes notice a message like:
!! No channel map, no channel, and no ds1? What am I supposed to
identify?
This is caused by a restart message from the switch containing no
channel ident IE. According to the ETSI standard this indicates a
restart of all B
harry gaillac wrote:
I agree you.
Does asterisk (Digium) project provide a good
documentation ?
[...]
If you think that all big IT corporations are virtuously advancing
technology for the benefit of us all, with no exception, and that the
OSS community is a bunch of worthless scums trying
Hi
all,
I'm using
asterisk 1.0.6 with bristuff-0.2.0-rc7k. I've already set up 3 boxes with
the same config, but I'm facing something strange with the fourth one
:
In my messages log,
I've got thoses lines :
kernel: zaphfc:
empty HDLC frame or bad CRC received (framelen = 4, stat =
Me thinks you haven't grasped the idea of OSS...
If you have a problem with it, do something about
it! (Dig in the project
and write some good documentation!)
I agree you I'll do it as soon as possible if i'm
allowed to post docs on asterisk.org .
I am however surprized you say there is
On 27/06/05, harry gaillac [EMAIL PROTECTED] wrote:
Why asterisk.org don't provide a documentation project
?
http://www.asteriskdocs.org/
--
Peter Bowyer
Email: [EMAIL PROTECTED]
Tel: +44 1296 768003
VoIP: sip:[EMAIL PROTECTED]
___
Asterisk-Users
Hello,
here is the SIP-log from my VOIP-phone when getting an MWI message from
asterisk:
NOTIFY sip:[EMAIL PROTECTED]:2054;line=g2kiz8tz SIP/2.0
Via: SIP/2.0/UDP 192.168.10.1:5060;branch=z9hG4bK33f02076
From: asterisk sip:[EMAIL PROTECTED];tag=as229cbc7c
To: sip:[EMAIL
I have noticed a strange behaviour when our internet connection was
down a couple of hours last week.
What happens is that asterisk starts running *really* slow. If I type
sip show peers it sometimes responds correctly and shows all the
connected peers, but sometimes I get an empty list (this
It is as simple as this: If you want everything for
free then sit on the
corner of the street with your hands open (called
begging).
Did i ask something for free ? Where I said I want all
for free?
Open source means you also have to do something for
it. You don't want to do
something
On Sun, 26 Jun 2005, Stefan Gofferje wrote:
Armin Schindler schrieb:
On Sat, 25 Jun 2005, Stefan Gofferje wrote:
Armin Schindler schrieb:
I have added busy()/congestion() support to CVS HEAD now, can you
please
test if it works for you?
Works
On Sat, 25 Jun 2005, Stefan Gofferje wrote:
Stefan Gofferje schrieb:
Armin Schindler schrieb:
I have added busy()/congestion() support to CVS HEAD now, can you
please
test if it works for you?
Works perfectly well! Also CallingPres(32) does work! The only thing I
Hi all,
on sourceforge.net I added the fixup release 0.5.2 of
chan_capi-cm driver.
The changes from 0.5.1 to 0.5.2 are:
- correct reset of channel pointer on hangup
(this should fix the seg-fault sometimes occured)
- added preliminary support for Busy()/Congestion()
Have fun
Armin
On 11:29:31 June 27, 2005 Armin Schindler [EMAIL PROTECTED] wrote:
I did some more testing.. On incoming calls, the caller hears the
called party very much chopped. On outgoing calls, the called
party hears nothing.
Using *1.0.7 bristuffed...
I cannot reproduce this here. Does it
On 11:28:09 June 27, 2005 Armin Schindler [EMAIL PROTECTED] wrote:
Yes, maybe you would like to implement some of those feature-requests
? ;-)
I would love to if I were a bright programmer :-).
--Stefan
--
(o_ Stefan Gofferje | Linux Systems Specialist
//\ Reg'd Linux User
Emanuele,
If you think that all big IT corporations are
virtuously advancing
technology for the benefit of us all, with no
exception, and that the
OSS community is a bunch of worthless scums trying
to make money off
other people's work, then please go buy those
enlightened firms'
I am attempting to get an iaxy device to connect to my
asterisk box over the public cloud however
It fails register and I cannot figure out why.
Below is my iax.conf, iaxy setup file and debug output
from iax2 debug.
My iax.conf
[u7402]
type=friend
accountcode=iaxy
host=dynamic
On 27/06/05, harry gaillac [EMAIL PROTECTED] wrote:
No i think time spent to work by developpers and users
merit a documentation project .
Please stop typing for a moment and start reading. There is a
documentation project for Asterisk at www.asteriskdocs.org.
Peter
--
Peter Bowyer
I had a similar problem with a recent install:
* TDM11B
* 1 Port HFC Card
I got that messages about HDLC Framing errors. It ended up being the
way I loaded the required kernel modules.
1) Remove or comment install lines in /etc/modprobe.conf (or
modules.conf), regarding the kernel modules
I have the same problem in a box with 2 HFC-PCI, but i already remove the row
in modprobe.conf and load the module manually.
Both cards works fine
Any idea ?
Giordano
-Messaggio originale-
Da: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Per conto di
Holy Cow. :)
Is there a patch for this ? Has a bug report been raised ? I couldn't
find any
Is there any work around for this until the bug is fixed ? Is it the
Restarting Channels that is aggravating the situation - I think that
there is an option to specify the restart timeout. I do
Hi all,
I'm trying to set up a test case for an ISP
featuring an asterisk server and a couple of linksys rt31p2-na routers
registering on it.
Instead of using dsl lines, i'm trying to plug the
* server and the routers on a cisco switch, just to test their
functionality.
I have created a
On Mon, Jun 27, 2005 at 11:04:17AM +0100, Julian J. M. wrote:
I had a similar problem with a recent install:
* TDM11B
* 1 Port HFC Card
I got that messages about HDLC Framing errors. It ended up being the
way I loaded the required kernel modules.
1) Remove or comment install lines
On Monday 27 June 2005 00:34, Jerry wrote:
Question: There are problems using FXS ports to pass data and get high
speed access, but can an FXO port be used as a modem? Nothing fancy
required, 12 or 2400 baud should be ok (9600+ would be nice, but practical
over nice). I know there is something
On Monday 27 June 2005 05:23, harry gaillac wrote:
I just told asterisk.org should provide a
documentation project
It's the job to all Asterisk developpers/users to
provide docs on Asterisk.org.
Wrong. The developers don't owe you (or anyone else) ANYTHING.
Either dig in and help out, or
On Monday 27 June 2005 02:45, Marcel van Kaam, Fonetica wrote:
I think by now everybody knows that LiveVoip went down, bankrupt etc
So please stop nagging about it and move on to some topics that really
matter.
If you want to discuss LiveVoip, get all together in a restaurant, eat,
drink
Are you pinging them from
the same VLAN? Or a VLAN that has access to it?
Marios
-Original
Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dionisis Koumouras
Sent: Monday, June 27, 2005 1:36
PM
To:
asterisk-users@lists.digium.com
Subject:
I have been investigating on this list and found that faxing is not
reliable between Zaptel
cards and that Digium does not support nor
recommend fax over the TDMXXB interfaces.
Is this true ? Will fax not be reliable enough for my customer ?
True... the TDM card (or its drivers)
On Mon, June 27, 2005 13:04, Andrew Kohlsmith said:
On Monday 27 June 2005 02:45, Marcel van Kaam, Fonetica wrote:
I think by now everybody knows that LiveVoip went down, bankrupt etc
So please stop nagging about it and move on to some topics that really
matter.
If you want to discuss
What's wrong with the standard 2.6 ztdummy?
It doesn't use RTC. I'm assuming you mean '1.0.8' as 'standard'.
How does HEAD zaptel interact with 1.0 asterisk?
Shouldn't cause any problems.
--Rob
___
Asterisk-Users mailing list
If you have Zaptel cards, does setting the build to USE_RTC use that
timing source in preference to the Zaptel card interrupts?
If you have a zaptel card, you shouldn't be loading ztdummy. So,
therefore, no problems!
--Rob
___
Asterisk-Users
Problem:
It seems the situation is improved when I remove the regsiter =
statements in my sip.conf.
Cause:
If your internet connection is down, your DNS isn't working. IF your DNS
isn't working, it won't be able to resolve names. If it can't resolve a
name, it will sit there trying until it
yes, I'm pinging the routers from the same
vlan.
Dionisis KoumourasS/W EngineerCyberStream
Ltd.tel. +30 210 729 7632fax. +30 210 729 7631mob. +30 694
7301104
___
Asterisk-Users mailing list
Asterisk-Users@lists.digium.com
i tried to write in routes, patterns to usa
destination 1.* it worked well but i wanted to specify
the number of digits then i tried 1NXXNXX but it
didnt work.
please help
__
Do You Yahoo!?
Tired of spam? Yahoo! Mail has the best spam
Question: There are problems using FXS ports to pass data and get high
speed access, but can an FXO port be used as a modem? Nothing fancy
required, 12 or 2400 baud should be ok (9600+ would be nice, but practical
over nice). I know there is something like this with the FAX extension
Hi all,
I was reading http://bugs.digium.com/view.php?id=2639 and it seems that
anthm's great native MoH patch only works on HEAD. Does anyone have a
version of the native MoH patch that works on 1.0.8? If so please point
me to its location or email it off-list.
Thanks and regards,
Patrick
Thanks for reply,
http://www.asteriskdocs.org/ is a good thing if i
could
contribute i'll do it .
However i don't find docs for all features asterisk
provide.
Why asterisk documentation is forked all over the web
?
anybody is free to write docs for any projects.
If digium host commercial
Hello All,
Ive been investigating and playing with asterisk to
see how it would work out as a small-medium business pbx to handle mostly
interoffice/branch communication and a possibly communication out to pstn in
later stages of implementation. (All communication would be VoIP
On Friday 17 June 2005 02:37, Chris Mason (Lists) wrote:
I don't want building1 to access international, but does it inherit
that include through including the office context? If it does, how
can I structure a dialplan so that each building can call each other
but building1 does not have
--- Andrew Kohlsmith [EMAIL PROTECTED]
a écrit :
Wrong. The developers don't owe you (or anyone
else) ANYTHING.
Why do you say you owe me ?
Tell me where I 've written that ?
Either dig in and help out, or move on. We won't
miss you. Frankly I get
really angry by this you owe me
snacktime wrote:
In case someone else made the same mistake I did, and because I can't
find this information posted anywhere, here is what I found out about
realtime sip.
You can use it to register UA's that are registering to asterisk, and
you can use it for peer context's for outgoing
Hello,
I am using asterisk with some isdn phones connected to a hfc based card
in nt mode, and one HT486 as sip-client. Everything is working so far,
but I want to achive this:
When a call is picked up on a phone (sip/zap should not make a
difference) there should be the possibility to put
Hi,
I have a Gateway running in TE (terminal equipment mode as slave that
I need to connect to my asterisk server using a TE100P card.
Can anybody give a quick run up of how to run the TE100P's in Network
Termination mode to have this working sucessfully?
Cheers!
Regards,
Sahil Gupta
Hello,
I saw some conversation about this in the archives, but nothing
definitive.
If a call comes in over a CO line via the TDM400P, the Comedian Mail
recording volume is so low it's inaudible. Calls coming in via SIP or
IAX do not have this problem.
Does anyone have any information on this
On 2005-06-26, Adam Megacz [EMAIL PROTECTED] wrote:
Rich Adamson [EMAIL PROTECTED] writes:
I've had pretty good luck with www.teliax.com
I like them too, except for support. I have THREE tickets open with
them that are ten days old and haven't received even a cursory we're
looking into it
I have a user who goes into Comedian Mail for the first time and goes
thru the initial setup, changes password, records name, etc. Problem is
that every time he calls in, it thinks that it's his first time and
keeps reprompting him. His password change is reflected in
voicemail.conf. Others do
These linksys boxes may be configured to ignore ICMP/PING
on the WAN side - so your connection may be ok! Go to the config web pages
of the router to enable PING.
OCG
From: Dionisis Koumouras
[mailto:[EMAIL PROTECTED] Sent: Monday, June 27,
2005 6:36 AMTo:
I think the $10 is setup, as you will notice all the others mention
the monthly next to the rate.
I was confused also. (Hint Teliax)
On 6/27/05, John Goerzen [EMAIL PROTECTED] wrote:
On 2005-06-26, Adam Megacz [EMAIL PROTECTED] wrote:
Rich Adamson [EMAIL PROTECTED] writes:
I've had pretty
We've had this before.
Go to the users vm directory (usually
/var/spool/asterisk/vm/mailboxnum), and remove all .wav, .WAV, .mp3 and
.gsm files
Then get them to re-record their message.
Julian.
Adam Robins wrote:
I have a user who goes into Comedian Mail for the first time and goes
thru
I just looked at their price page. Each package says setup is Free.
Now, I do notice that the Price for the pay as you go doesn't have
/mth. on it as the others do. So maybe there is a difference. I
agree with you that it is not extremely clear and they could do a whole
lot better job
Hi,I'm writing an AGI script to manage outgoing calls. We need to interrogate a database to work out which line a particular user is allowed to use for outgoing calls. However, I cannot find a way for my AGI script to access the SIP username. Does anyone know if this is possible (even if it is
On Monday 27 June 2005 08:05, harry gaillac wrote:
Why do you say you owe me ?
Tell me where I 've written that ?
It's the job to all Asterisk developpers/users to
provide docs on Asterisk.org.
That is where you have essentially written You have provided the code to me
for free, now it's also
On Monday 27 June 2005 08:27, John Goerzen wrote:
I'm looking for someone that sells minutes in bulk like LiveVoip used
to. No monthly fee, just pay-as-you-go. It looks like Teliax charges a
minimum of $10/mo, even if I use no minutes that month.
http://www.nufone.net. I've been using them
On Monday 27 June 2005 07:21, Francesco Peeters wrote:
*Shrugs* Seen it, been there... This happens on all lists at some point
in time... Several lists I am on have already created an OT or TALK list
besides the main list...
Whenever that happens, I just subscribe to the 2nd list and
Hello,
Why dont you try using
the the CALLERIDNUM or maybe the ACCOUNTCODE and based on that to take your
decision?
-Original
Message-
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of David Shirley
Sent: Monday,
June 27, 2005 4:52
PM
To: Asterisk Users Mailing
maybe its a minimum amount? Haven't signed up so I don't know.
On 6/27/05, Dan Perik [EMAIL PROTECTED] wrote:
I just looked at their price page. Each package says setup is Free.
Now, I do notice that the Price for the pay as you go doesn't have
/mth. on it as the others do. So maybe there
I have been trying to configure a similar configuration (a digium card
connected to a mobile phone, in order to make calls to the mobile network,
and a HFC ISDN card, connected to two lines, the 2 ISDN cards).
In order to make this configuration I don't know to make the configuration
(I have
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Marcel van Kaam, Fonetica
Sent: Sunday, June 26, 2005 11:46 PM
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
Subject: RE: [Asterisk-Users] LiveVoip is Bankrupt
Hi All,
I think
Hello..
How is this possible?? I have 65 active calls .. but
making new calls and registering isn't possible
anymore
the CLI command restart now didn't even work .. had to
kill the process before it worked
again...
myasterisk*CLI show
channels Channel
(Context Extension Pri )
I am attempting to get an iaxy
device to connect to my asterisk box over the public cloud
however
It fails register and I cannot
figure out why.
Below is my iax.conf, iaxy
setup file and debug output from iax2 debug.
My
iax.conf
[u7402]
type=friend
accountcode=iaxy
host=dynamic
Has anyone written a
LogWatch script for Asterisk? I use logwatch for monitor all my critical
services and would like to do the same for Asterisk.
LogWatch is very
popular, so I'm guessing that someone has created one but hasn't had time to
post it somewhere...
Thanks,
OCG
I'd be happy to host you in our Montreal Datacenter at no cost.
Contact me off-list if you're interested.
/Danny
Matt Riddell wrote:
Andres wrote:
So it looks like Livevoip went Bankrupt
Sh1t.
Looks like the Daily Asterisk News will need a new host.
So, unless anyone can donate space
On 27/06/05, harry gaillac [EMAIL PROTECTED] wrote:
May be it would be simplest to search for help in docs
on asterisk.org .
Anybody is free to write a plethora of
documentation.
The state of Asterisk documentation certainly isn't ideal. But
equally, the way to improve it isn't to post here
For the pay as you go plan, there is no set up fee. The $10 they have
listed is the minimum increment you buy minutes with. When you first sign
up, you pay $10 and get $10 worth of minutes. Then you can manually pay
more to add to your account balance or use there auto replenish setup. The
i guess you have to trust the agi_callerid and agi_channel var. With
agi_channel you know if the originating party is a SIP, then with the
caller id you know the sip user.
best regards.
On 6/27/05, David Shirley [EMAIL PROTECTED] wrote:
Hi,
I'm writing an AGI script to manage outgoing calls.
Hi Andrew,
Andrew Kohlsmith wrote:
On Monday 27 June 2005 00:34, Jerry wrote:
Question: There are problems using FXS ports to pass data and get high
speed access, but can an FXO port be used as a modem? Nothing fancy
required, 12 or 2400 baud should be ok (9600+ would be nice, but
practical
Ah thanks Nicolás.
I will give it a try and let you know how it goes.
Kindest regards
David Wilson
___
D c D a t a
Tel +27 33 342 7003
Fax +27 33 345 4155
Cell +27 82 4147413
http://www.dcdata.co.za
[EMAIL PROTECTED]
Powered by Linux, driven by passion !
On 6/26/05, harry gaillac [EMAIL PROTECTED] wrote:
Hello,
I read
http://www.voip-info.org/tiki-index.php?page=Asterisk+RealTime+Voicemail
however when i run show voicemail users app voicemail
return users in voicemail.conf
Why?
You should enable debugging in the console (logger.conf)
And fee's for the Tollfree numbers.
We us them also and have had a good experience.
Rick
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
[EMAIL PROTECTED]
Sent: Monday, June 27, 2005 7:49 AM
To: Andrew Latham; Asterisk Users Mailing List -
I would be interested in this also.
Thanks
Rick
From:
[EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Michael Stahl
Sent: Monday, June 27, 2005 7:32
AM
To: Asterisk
Users Mailing List - Non-Commercial Discussion
Subject: [Asterisk-Users] LogWatch
for Asterisk
You just got a tax write off because your money is certainly locked up
in chapter 11.
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Bashir
Ullah - www.Lamsre.Com
Sent: Sunday, June 26, 2005 4:02 AM
To: Asterisk Users Mailing List - Non-Commercial
As predicted
In keeping with LiveVoip company policy, even this letter seems
antagonistic towards customers and creditors. You are under a STAY!!
Don't talk to us!
Wow, I guess the merger with the trailer park DSL company just did not
help at all. And after Joop spent so much time
Well, I guess stating at the bottom of the list is a bad idea sometimes.
Sorry, Marcel, I just find this a relevant topic since so much money and
time have been wasted trying to use this company's service.
Will drop it shortly though.
Cheers,
W
-Original Message-
From: [EMAIL
Actually, if you look at my posts from a month or two ago, you can see
that they not only had to have known, they were publicly stating that
they were expanding. Joop personally told me that they were going to
offer Vonage type of service and that they were opening service in the
UK. He actually
For the record/archives, I tested with asterisk sending calls to the
MAX with the Display IE in the SETUP message and the MAX still doesn't
send the name in headers. Another helpful soul pasted me some config.
I'll keep at it and report my findings here, at least for the bots to
index.
My
Andrew Kohlsmith wrote:
On Monday 27 June 2005 02:45, Marcel van Kaam, Fonetica wrote:
I think by now everybody knows that LiveVoip went down, bankrupt etc
So please stop nagging about it and move on to some topics that really
matter.
If you want to discuss LiveVoip, get all together in a
Trying to find some answers regarding
Asterisk and Voice Bridging.
Mainly, how robust is it? Are
you able to assign user and admin passcodes?
Also, if the admin is not logged in
to the conference, will the users be able to log in?
and if so, will the users be able to
hear talk to each other?
LiveVoip has been a learning experience for anyone who purchased from
them. With any luck, it was a learning experience in what not to do
for anyone out there that provides similar services. At least I hope so
since it seems obvious that LiveVoip never learned a thing during their
interaction
Having read the TOS from LiveVoip many times, I can almost assure you it
was written by the LiveVoip staff and not a lawyer. Due to that, I
cannot imagine them slithering out of this entirely.
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Marie
I solve my problem using stable version of zaptel and CVS HEAD of asterisk, asterisk-addons,libpri
and in two days of testing work well.
On 6/22/05, Paradise Dove [EMAIL PROTECTED] wrote:
I have the same problem.seems that tdm400b is not working on CVS HEADOn 6/18/05, Steve Totaro [EMAIL
Ok,
I need help to configure voicemail conf and messages
in db mysql via ODBC and ARA .
I read and googled
http://www.voip-info.org/tiki-index.php?page=Asterisk+RealTime+Voicemail
without success.
Thanks for help
harry
--- Peter Bowyer [EMAIL PROTECTED] a écrit :
On 27/06/05, harry gaillac
On 6/27/05, Steve Blair [EMAIL PROTECTED] wrote:
snacktime wrote:
In case someone else made the same mistake I did, and because I can't
find this information posted anywhere, here is what I found out about
realtime sip.
You can use it to register UA's that are registering to
Hello,
I'm having trouble getting finding a company that
provides DID in Western Canada. More specifically in
Edmonton, Alberta.
We have tried getting in contact with Link2Voip and
Calgary Telecom but neither seems to be answering
their phones or email.
I would appreciate it if anyone can
I was charge $25 for a toll free port. I don't know what the fee is for a
local port, though I have one in progress. I haven't been able to find
these fees listed on the website (yet). Maybe someone from Teliax can fill
in what extra fees may be charged?
Doug
At 10:17 AM 6/27/2005, you
Hi,
can someone here explain in detail what exactly the dropcount value
represents and how it is calculated? From the help I can read the value
of 3 represents of 1.5% frames dropped. What value of 5 means? Is it
reasonable to set this to higher value, like 10?
Thank you,
--
-
David Hajek
This is probably a good time to point out that there is a good litmus
test for all Voip providers. PRIOR to purchasing anything, send them an
email and request the sales information. Ask about their servers or
their policies or anything you can think of. How they respond will tell
you a lot.
Andrew Kohlsmith,
Can you tell me why a documentation project on
asterisk .org is a serious problem for you and others
people ?
When i say It's the job to all Asterisk
developpers/users to provide docs on Asterisk.org.
Job mean in order to help people
That is where you have essentially
While using Blindtransfer #Extension
everything works fine.
But how do i activate announced transfer
with an Grandstream GPX2000 ?
Greets
Markus
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On 2005-06-27, Andrew Kohlsmith [EMAIL PROTECTED] wrote:
http://www.nufone.net. I've been using them for the past 18 months with zero
technical hassle. Jerjer and Shido6 hang out on IRC. Nufone is not a hand
holding VOIP provider. You are expected to have some clue. This has turned
Hi Wiley,
I understand that, and it was not my intentions on being to negative.
It is just that this is the users list and for the Biz we have the biz list.
Here on the users list we should discuss topics about using asterisk and on
the biz list we should discuss offers and bankruptcies.
I
Well, as someone who doesn't use threads... I think I can say it is not
the end of the world. I find scanning my Asterisk mail folder to be
pretty easy
W
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Matt
Riddell
Sent: Monday, June 27, 2005 8:47
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