What is a RMA?
Isamar Maia escribió:
Just for curiosity, Let us know how much time you'll gonna get a RMA of
it.
Isamar
On Thu, 22 Jul 2004, Andres Junge wrote:
I had the same problem, and it was that the power suppply coudn't handle
the new card. My solution (until i get a new power supply)
Hello I am new to asterisk I want to setup the call queues where it will
ring multiple devices at the same time and send the call to the first one
that is picked up. There doesn't need to be an agent login for this I don't
think I just want setup so no login is required. Please help
-Jeremy
Hello from Spain,
I have in my company a PBX that we are using at this moment. We want to
migrate to VoIP, so after investigating for about a week we decided to use
Asterisk, but save the old PBX. We have 7 telephone lines, and we want to
connect 7 extensions from the PBX to the Linux box used
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
I've confirmed that the stack overflows caused by the X100P driver are
caused by running it in a 4K-stacks kernel - it works ok with the linuxant
16K-stacks kernel, even with the CPU under a high load.
- --
- Steve Jabber: [EMAIL
And I would very much like feedback on those - are they useful?
If they are, I'll backport to chan_sip.
I find them useful (for the HTML based block LED field display of DeStar).
Today I even thought about writing a patch that sends the
available/unavailable messages from the quality=... code
Robert Jackson wrote:
-Original Message-
From: John Angelmo [mailto:[EMAIL PROTECTED]
Sent: Thursday, July 22, 2004 11:10 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users] Faild Echotest
But when I call the echotest it just hangs up, echotests from
other VoIP
Are you answering the
And I would very much like feedback on those - are they useful?
Oh, I just found out by looking at the source code that there are database
entries SIP/Registry. I think the used database entries is something that
is currently under-documented ... :-)
At 9:41 AM +0200 on 7/23/04, Holger Schurig wrote:
And I would very much like feedback on those - are they useful?
If they are, I'll backport to chan_sip.
I find them useful (for the HTML based block LED field display of DeStar).
Today I even thought about writing a patch that sends the
Looking for firmware (anything) for the 12sp model
phones. Anyone got it drop me a line!
[EMAIL PROTECTED] or
[EMAIL PROTECTED]
Sincerely,
Steve McMahonDigital DataBits InnovationsSalem, OR
97301Office: (503)371-6448 Ext. 2Cellular:
(503)881-6828
Holger Schurig wrote:
And I would very much like feedback on those - are they useful?
Oh, I just found out by looking at the source code that there are database
entries SIP/Registry. I think the used database entries is something that
is currently under-documented ... :-)
If you simply type
The patch tries to send the time as well, but it fails. There are some
problems currently:
I applied the path from bug 2117. After this I got some events:
Event: SIPPeerStatus
Peer: weckhardt
Status: reachable
Time: 55
Event: SIPPeerRegistration
Peername: dnarotam
Status: Offline
I think we
Hi!
What I also saw in my little research that * in not suitable for
large deployments like medium or large enterprises or sometimes even
smalls ones with specifics needs, any of you could mention a lot of
them, Call Centers, Banks, etc. and why not carriers for their own use.
Just a
Hi,
-Original Message-
Subject: [Asterisk-Users] Cisco 12sp firmware... Anyone got it???
Looking for firmware (anything) for the 12sp model phones.
Anyone got it drop me a line!
You're stuck with what's on them. There is no firmware upgrade for 12SP
AFAIK (besides, they are -and
I've tried setting nat=yes in places, externip, et al with no success ..
even though the code I was running from back then worked without that.
Some of the options in sip.conf have changed look at the samples in
src/asterisk/configs/sip.conf.samples
Regards
Jason
Okay, I have finished my patch. With qualify=yes in sip.conf it looks
like this:
-
# telnet 127.0.0.1 5038
Trying 127.0.0.1...
Connected to localhost.localdomain (127.0.0.1).
Escape character is '^]'.
Asterisk Call Manager/1.0
Action: Login
Hi Jeremy,
What about this in extentions.conf
exten = 5000,1,Dial(SIP/phone1SIP/phone2SIP/phone3,50,r)
--
mvh. Hans-Henrik Andresen
Jeremy Kenney [EMAIL PROTECTED] wrote in message
news:[EMAIL PROTECTED]
Hello I am new to asterisk I want to setup the call queues where it will
ring
On Fri, 23 Jul 2004 02:26:26 -0400, Jeremy Kenney [EMAIL PROTECTED] wrote:
Hello I am new to asterisk I want to setup the call queues where it will
ring multiple devices at the same time and send the call to the first one
that is picked up. There doesn't need to be an agent login for this I
- Original Message -
From: Steve McMahon [EMAIL PROTECTED]
Date: Fri, 23 Jul 2004 01:12:26 -0700
Subject: [Asterisk-Users] Cisco 12sp firmware... Anyone got it???
To: [EMAIL PROTECTED]
Looking for firmware (anything) for the 12sp model phones. Anyone got
it drop me a line!
Holger Schurig wrote:
The patch tries to send the time as well, but it fails. There are some
problems currently:
Seems like you are getting Time - how does it fail? Please explain...
/O
I applied the path from bug 2117. After this I got some events:
Event: SIPPeerStatus
Peer: weckhardt
Status:
Holger Schurig wrote:
I think we have several problems here. Once it's Peer:, the other time
it's Peername.
That's clearly a bug.
Also, I don't like the name of the event. It should just
be PeerStatus and PeerRegistration, because we might add something to
IAX2 as well. So I'd suggest to do
Hi,
I've never run against a commercial PBX that didn't need
maintenance.
Acknowledged.
VM hard
drives fail,
...
Asterisk is
every bit as stable
as the old-gen KSUs and PBXs.
There are big differences. As I know of no other PBX that uses 'consumer' hardware,
asterisk is also
Holger Schurig wrote:
And while I was at this patch, I also changed the
Event: SIPRegistry
Domain: ...
Status: ...
to
Event: Register
Channel: SIP
Domain: ...
Status: ...
I still believe it would be better to call this Registry since that's a common
term across IAX and SIP for outbound
Quoting Jeremy Kenney [EMAIL PROTECTED]:
Hello I am new to asterisk I want to setup the call queues where it will
ring multiple devices at the same time and send the call to the first one
that is picked up. There doesn't need to be an agent login for this I don't
think I just want setup so no
Quoting Philipp von Klitzing [EMAIL PROTECTED]:
What I also saw in my little research that * in not suitable for
large deployments like medium or large enterprises or sometimes even
smalls ones with specifics needs, any of you could mention a lot of
them, Call Centers, Banks, etc. and
Hi there,
I'm having problems with the Grandstream Budgetone 101 on hangup -
show channels/show channels concise output is still showing the
call's channels as active.
The problem does not exist when I use SJPhone, so I'm assuming it isn't
an Asterisk configuration issue. Has anyone seen this,
Hi,
is there any mean to get ringing tones for the
E100P, according the signals from the telco?
When I use the r-option in extensions.conf, I have
too early ringing tones, i.e. when busy, I have
ringing tones for a while and afterwords the busy
tone.
If I don't use the r-option I have no tone at
Hi Michael,
You were right. When I applied a patch to openh323, it solved my problem
Thank you Michael.
MandarMichael Manousos [EMAIL PROTECTED] wrote:
Try to describe your problem. A first guess is that you didn'tapply the patch for the OpenH323.Michael.Mandar Pise wrote: Hi Folks, I am
I hate to disappoint anybody, but we're having AstriCon at
the Atlanta Marriott Century Center on the north side of the
city just outside the Buckhead entertainment district.
The Century Center is a different property from the Marquis.
If we had held AstriCon at the Marquis, the event would
Hi,
I am using SIP extensions connected to the PSTN with the CAPI Channel
driver.
All works fine except that one of the sip phones keeps ringing when the
caller
hangs up before extension is answered. The phones are grandstream 100,
though
we get the same behaviour using other phones (X-lite,
Avizion, you're joking right?
-= Info about application 'AddQueueMember' =-
[Synopsis]:
Dynamically adds queue members
[Description]:
AddQueueMember(queuename[|interface[|penalty]]):
The AddQueueMember function does indeed allow you to set the penalty.
Too bad penalties don't work
Why is no one suggesting any solution here for this problem, which has been lingering
for a while.
Are we not supporting H.323 on Asterisk?
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of ruixun wu
Sent: Thursday, July 22, 2004 4:06 PM
To: [EMAIL PROTECTED]
hi
I tried to answer the * poll, but...
Warning!
This free account has reached its monthly entry limit
No more entries are accepted at this time.
roy
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
John Angelmo wrote:
I tried that but I still get:
-- Executing Answer(SIP/2000-00b8, ) in new stack
-- Executing Echo(SIP/2000-00b8, ) in new stack
== Spawn extension (from-sip, 700, 2) exited non-zero on 'SIP/2000-00b8'
dev*CLI
I can now say that the error occurs everywhere I try to
Thanks, I _finally_ got unixODBC and FreeTDS
working with MSSQL. I hate to through all that hard work out the door, but
I like your idea better.
Is it in cvs now, ready to go? I read that
mark was waiting on a fix... ?
Thanks for the link.
Thanks
Duane Cox
- Original Message -
On Wed, 21 Jul 2004 19:34:41 -0500, Steven Sokol
[EMAIL PROTECTED] wrote:
Guys, the hotel costs are $111.00 including tax per night. That is actually
very reasonable for a decent hotel. I'm willing to bet that the place you
found for 1/2 the cost of the Marriott is not capable of hosting a
The last post I saw regarding this was June 2003 (before my time with *, I
started in September):
http://lists.digium.com/pipermail/asterisk-users/2003-June/013324.html
Is there any support for Q.SIG in * at all? The Norstar MICS uses a MCDN
(Meridian Customer Defined Network) key to enable
Hello,
avizion wrote:
This is exactly what I will be looking for in near future. Our current setup
(Old Ericsson PBX) with these system phones having hotkeys for transfer,
hold, ACD in/out, multiple lines, etc. and a quite handy feature... the LED
that tells my weather a certain agent is busy or
Hello,
I am running asterisk on the same machine that acts as my nat-router and
firewall to the internet.Its connected via DSL and works quite fine.
The machine is registered to famous (?) german sipgate.de via a sip-account.
This works. But when I (or somebody else) in the network starts some
PS: If already existing soft (and/or hard) phones have more of this
functionality - please let me know.
WAMi and other gui interfaces already support this.
http://www.voip-info.org/wiki-Asterisk+WAMI
We are starting work on WAMi 2.0, and I am trying to make the source
available for everyone
List,
A correction. This problem has been fixed by us by applying the H.323 patch.
Thanks Mandar. Good work
Seshu Kanuri
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Kanuri, Seshu
Sent: Friday, July 23, 2004 9:09 AM
To: [EMAIL PROTECTED]
Subject:
hehe - well - I think you missed the word persistant :)
After a few hours of digging in old docs - I also found the new 3rd parameter
in a more recent doc. Not sure how I missed in the first place - but I did.
Never the less it did not solve my problem of having members added to my queues
I'm trying the latest bri 0.1.0 RC2 drivers.
In announce I see implementation of so long waited Transfer feature.
But I can't make it work.
When the person who is making transfer after talking with second party press
R second time to establish 3 way call
the person to which call supposed to be
This works. But when I (or somebody else) in the network starts some heavy
Just forgot to say:
The problem is the outgoing traffic, when there is download for example via
ftp, the other end can hear me without problems. But when there is upload to
somewhere, everything stutters.
--
--
Sterkel
I had have this problem before.
Just try.
1. PWLIBDIR=put here path wher you have pwlib.
(I have located my pwlib in /root/pwlib/ and I have PWLIBDIR=/root/pwlib
2. export PWLIBDIR
3. OPENH323DIR=put here path wher you have openh323
4. export OPENH323DIR
5.
Just a followup so this stuff stays in the archives:
Cisco's documentation on q.sig:
http://www.cisco.com/univercd/cc/td/doc/product/software/ios120/120newft/120t/120t2/qsigfm.htm
Basically q.sig is modelled after q.931. It might then be possible to get *
to handle q.sig natively without too
It is the hard phones that need this before Asterisk is a salable
solution to small/medium businesses. What sells the system is the
phones and the flashing lights.
As most users already have a legacy system with a real BLF etc, until
Asterisk has hard phones that have all those features it will
Hi people.
I've been having some audio problems with some of my cisco 7940 phones
using firmware 7.1. The sound gets gargoled up, I can't understand much
of what is said (listing on the IP phone).
My setup is the following:
7940 - * - Cisco 2621(GW) - T1
I'm using SIP and g729 for all stages
This works. But when I (or somebody else) in the network starts some heavy
Just forgot to say:
The problem is the outgoing traffic, when there is download for example via
ftp, the other end can hear me without problems. But when there is upload to
somewhere, everything stutters.
try to
I have a D-Link 1120M MGCP Telephone Adapter with Asterisk
using MGCP. Call setup / connects work ok. Can call from a fxs
phone and can receive calls to the fxs phone. The problem is there is no
audio to 1120M fxs phones. There is audio to the far end device (SIP and
IAX). I have collected
Googling for QoS linux packet shaping gives many auspicious results
that may be worth looking at, also.
Chris.
Senas Jordanovic wrote (on Jul 23):
This works. But when I (or somebody else) in the network starts some heavy
Just forgot to say:
The problem is the outgoing traffic, when there
On Fri, Jul 23, 2004 at 09:44:38AM -0400, Andrew Kohlsmith wrote:
Is there any support for Q.SIG in * at all? The Norstar MICS uses a MCDN
Asterisk can do basic things w/ Q.SIG. Q.SIG is a LOT like q.931 in basic
call handling, etc. The part where Asterisk lacks is in a lot of the
detailed
Tony Nichols wrote:
On Mon, 2004-07-19 at 13:36, Christopher L. Wade wrote:
Exactly which NEC T1 interface did you use? I'm looking at the DTI-U20,
I don't think I'll need the U30, but I'm not entirely sure.
Thanks,
Chris
I used the DTI-U10 (DTI-24-U10). Got it from GTS Telephone
Hello,
I recently tried an upgrade of CVS on my test server today and found that
the res/res_parking.c file is completely gone. This is where I had to go
into the code every time I do an upgrade and change the code to allow for
doublehash transfers instead of single hash transfers:
That
I have a little menu set up where hitting 1, 2, or 3 places the call through
to a cellular phone over IAX. That works. However, if caller hits 4 to go
into voicemail, the system hangs up. Am I doing something wrong in the dial
plan, or is this a CVS change? I had no trouble with this until I
Hi Michael,
I do check the mail list for this problem. But all
the answer is using incorrect libraries at run time.
And didn't give more details. I just don't know how to
use the correct libraries at run time.
Now I have installed Pwlib 1.6.6 and Openh323
1.13.5, why do you say I am using
On Fri, 23 Jul 2004 12:00:22 -0400, mattf [EMAIL PROTECTED] wrote:
That means that you need to hit the pound key twice to initiate a
transfer instead of once. Because of our inbound call center we need to do
transfers and we also need to be able to hit the pound key once without
At 12:00 PM -0400 on 7/23/04, mattf wrote:
Hello,
I recently tried an upgrade of CVS on my test server today and found that
the res/res_parking.c file is completely gone. This is where I had to go
into the code every time I do an upgrade and change the code to allow for
doublehash transfers
Hi,
I would like use codec G7231A6K3 with oh323, but seems asterisk don't
undestood this codec.
I can't use G7231, the remote gateway don't accept this version of G723.
Thanks for help.
--
Arnaud Pignard ([EMAIL PROTECTED])
Frontier Online - Opérateur Internet
At 10:58 AM +0200 on 7/23/04, Holger Schurig wrote:
Okay, I have finished my patch. With qualify=yes in sip.conf it looks
like this:
-
# telnet 127.0.0.1 5038
Trying 127.0.0.1...
Connected to localhost.localdomain (127.0.0.1).
Escape character is
Hi Seshu,
It's great to hear someone fixed the problem.
By the way, could you tell me how to apply the H.323
patch?
Thanks a lot
Rui
--- Kanuri, Seshu [EMAIL PROTECTED]
wrote: List,
A correction. This problem has been fixed by us by
applying the H.323 patch.
Thanks Mandar. Good work
John Todd wrote:
At 12:00 PM -0400 on 7/23/04, mattf wrote:
Hello,
I recently tried an upgrade of CVS on my test server today and found
that the res/res_parking.c file is completely gone. This is where I
had to go into the code every time I do an upgrade and change the
code to allow for
Read README in oh323 directory, use exactly libraries you can read there,
and obviusly apply patch first...
Then run ldconfig
Put variables on environment
And all is ok
-Mensaje original-
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de ruixun wu
Enviado el: Viernes, 23 de
John Todd wrote:
I hate being a me too poster, but the double-hash patch I have
implemented four times now, and I know at least three other people who
have also gone well out of their way to put that patch into their
system. Making this an official modification would be ideal, in my
opinion,
Hi!
I've bought a Wildcard X101P card to play a bit with
asterisk.
I'm just trying to place a call to a number, wait
until the receiver takes the call, and then play one
of the .gsm sound files.
I'm in France, so first I tried modprobing wcfxo with
opermode=1, but the DAA mode remained FCC, so
Asterisk doesn't support any form of G723 except with Pass through... You
might try G726 or G729
Thanks, Billy
- Original Message -
From: Arnaud Pignard [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, July 23, 2004 12:45 PM
Subject: [Asterisk-Users] oh323 codec G7231A6K3
Hi,
-- GOOD NEWS! --
As ground transportation and the costs associated therewith are something of
an issue. So, in the spirit of making AstriCon available I've contacted the
hotel and arranged/discovered some low cost options:
-- MARTA --
MARTA, the Atlanta mass-transit system (i.e. train) can get
Hello,
I have some reports from users that occasionally DTMF will stop working in
voicemail and they will have to exit the system to get it to work again.
The useragents are Polycom IP 500's and I am using dtmfmode=rfc2833, with
Ulaw codec. This is all on an internal switched 100mb lan.
Has
On Fri, 23 Jul 2004, Brent Franks wrote:
Hello,
I have some reports from users that occasionally DTMF will stop working in
voicemail and they will have to exit the system to get it to work again.
The useragents are Polycom IP 500's and I am using dtmfmode=rfc2833, with
Ulaw codec. This is
Hi,
I'm trying to set up a basic FXO SIP gateway. That is, I want calls
from my SIP phone to simply be dumped onto the POTS line. My (entire)
extensions.conf is:
[from-sip]
exten = _9NXX,1,Dial(ZAP/1/${EXTEN})
and my zaptel.conf is:
fxsks=1
loadzone=us
Hi Steffen,
I'm sorry about my late response!
Thank you very, very(!!!) much, I had to learn how to handle the patch you
send me, but now it works great!
I changed the line in the chan_sip.c from stable V1.0 and it works great!!!
old: add_header(resp, Contact, contact);
new:
Hi,
I am a new user to both Linux and Asterisk and would be grateful for any help
and advice anyone has to offer. I have installed Linux and asterisk as per Andy
Powells excellent getting started guide. The problem I have is that the
addmailbox utility does not work and I cannot find the
On Fri, 2004-07-23 at 11:51, Christopher L. Wade wrote:
Tony Nichols wrote:
On Mon, 2004-07-19 at 13:36, Christopher L. Wade wrote:
Exactly which NEC T1 interface did you use? I'm looking at the DTI-U20,
I don't think I'll need the U30, but I'm not entirely sure.
Thanks,
Chris
On Fri, 2004-07-23 at 14:15, neil wrote:
The problem I have is that the addmailbox utility does not work and I
cannot find the file anywhere on the machine.
There no need to manually use the addmailbox script (it's in
/usr/src/asterisk/contrib/scripts by the way). As long as you define
your
Does anyone know which signalling protocol works when connecting the
output of a NT8B90AL Norstar ATA2 analog terminal adapter to an FXO
card? (fxsks, fxsgs, fxsls?)
I'm trying to add an Asterisk branch to my Norstar PBX as outlined at
http://www.voip-info.org/wiki-Asterisk+Nortel.
- Mike
Just to keep the list/archives up to date.
MCDN is not Q.SIG. The Norstar systems can do Q.SIG but they need to be
configured as NON North-American in order to get the option to show up (and
possibly require the use of E1 signalling now instead of T1). None of this
really matters with *
On Thu, 2004-07-22 at 15:58, Chris Shaw wrote:
As for multiple lines, they do offer multiple virtual numbers but unless you
want it to look like you have multiple lines, you don't need to do that...
VoIP by nature supports multiple calls to the same phone number without the
need for Trunk
On Thu, 2004-07-22 at 21:35, Lion Templin wrote:
I've been using * CVS code from May of this year and was able to connect
to iConnectHere and receive calls with * being behind NAT. Now that
I've upgraded to 1.0 RC1, this no longer works.
I've tried setting nat=yes in places, externip, et
The ata on a norstar is for connection of analogue devices and not trunks so
the i/f is loop start fxs
Neil
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Dr. Michael J.
Chudobiak
Sent: 23 July 2004 19:55
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users]
At 6:59 PM +0200 on 7/23/04, Olle E. Johansson wrote:
John Todd wrote:
I hate being a me too poster, but the double-hash patch I have
implemented four times now, and I know at least three other people
who have also gone well out of their way to put that patch into
their system. Making this an
Some unsuccessfull attempts to make console calls working.
If a sip phone is called, the other side will hear nothing.
If I try to record some sound the application will not finish. There
is a sound file, but it is empty (0 bytes). Record(${FILE}:gsm|10|30|skip)
is used in the dial plan. After
Hi John,
John Todd wrote:
At 10:58 AM +0200 on 7/23/04, Holger Schurig wrote:
Okay, I have finished my patch. With qualify=yes in sip.conf it looks
like this:
output snip
Event: PeerStatus
Peer: SIP/weckhardt
PeerStatus: Reachable
Time: 81
some more snip
On Wed, 7 Jul 2004 12:57:01 +1000, Shaun Ewing [EMAIL PROTECTED] wrote:
BTW: We can *always* use more help documenting...
Good point.
My C knowledge is very basic (ie: I could write basic programs, and
make simple modifications, but that's about it), so I can't contribute
that way, but
There shouldn't be an overage rate though if you're on the unlimited plan
like I am...
- Original Message -
From: Dameon D. Welch-Abernathy [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, July 23, 2004 12:33 PM
Subject: Re: [Asterisk-Users] VSP? Looking for advice.
On Thu,
Hi Alexey,
I followed your steps, but Asterisk still didn't
work. I am a little crazy. I show my envirement and
ld.so.conf here. Could somebody tell me if I am using
the correct libraries?
Thanks a lot
ld.so.conf:
/usr/kerberos/lib
/usr/X11R6/lib
/usr/lib/qt-3.1/lib
/usr/local/lib
Hello,
Well it gets me half way there, pressing ## does work with this patch, but
pressing # once doesn't send a hash tone out, it gets muted while it waits
to hear another tone. any way around this so you can press a single hash and
have it send a tone through? That's the way the original
Googling for QoS linux packet shaping gives many auspicious results
that may be worth looking at, also.
I come closer maybe, but this RTP is suspicious and its not helping to
priorize just the port 5060 I think.
--
--
Sterkel
FAX: +49-180 533 31 60 9328
+++ GMX DSL-Tarife 3 Monate gratis*
On Fri, 23 Jul 2004, John Todd wrote:
I think I've talked about this a few times in a few different posts,
but the short of it is: wouldn't it be great to just press a single
key to have some dialplan action taken without hanging up on either
side of the conversation? This should
On Fri, 2004-07-23 at 12:50, Chris Shaw wrote:
There shouldn't be an overage rate though if you're on the unlimited plan
like I am...
Not according to the CEO of BroadVoice:
http://www.voxilla.com/voxstory71-nested-order0-threshold0.html
-- PhoneBoy
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
On Friday 23 July 2004 12:21 pm, Brent Franks wrote:
Hello,
I have some reports from users that occasionally DTMF will stop working in
voicemail and they will have to exit the system to get it to work again.
The useragents are Polycom IP 500's
Can HTB minimize latency better than CBQ?
Just curious,
Rich
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Senas Jordanovic
Sent: Friday, July 23, 2004 1:00 PM
To: [EMAIL PROTECTED]
Subject: RE: [Asterisk-Users] Priorizing of packets
This
I have been a reseller subscriber of pipecall since
they started, however I am really struggling to get pipecall to work for
outbound or inbound calls. I get errors that the registration has timed out.
I have tried many variations of the register command
register = [EMAIL
I also want this to drop into the queue will that do this or do I need to
add something to queues.conf
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Hans-Henrik
Andresen
Sent: Friday, July 23, 2004 4:58 AM
To: [EMAIL PROTECTED]
Subject: [Asterisk-Users]
Dameon D. Welch-Abernathy wrote:
On Thu, 2004-07-22 at 21:35, Lion Templin wrote:
I've been using * CVS code from May of this year and was able to connect
to iConnectHere and receive calls with * being behind NAT. Now that
I've upgraded to 1.0 RC1, this no longer works.
I've tried setting
Subject: Re: [Asterisk-Users] Asterisk + NEC Electra Elite IPK Integration
Reply-To: [EMAIL PROTECTED]
I haven't really gotten too far into this, but I was wondering just what
'features' of the NEC phones (DTH-16D-1(BK)TEL) I'll be able to work
with from *? I'm currently getting some
Sterkel Brandke wrote (on Jul 23):
Googling for QoS linux packet shaping gives many auspicious results
that may be worth looking at, also.
I come closer maybe, but this RTP is suspicious and its not helping to
priorize just the port 5060 I think.
Prioritise based on source/destination of
Not according to the CEO of BroadVoice:
http://www.voxilla.com/voxstory71-nested-order0-threshold0.html
-- PhoneBoy
Hmmm... Well I'd be careful then... I've tried it several times and I
haven't gotten charged, but then again I'm not doing it from multiple IP
addresses either, it's all from
- Original Message -
From: Dr. Rich Murphey [EMAIL PROTECTED]
To: [EMAIL PROTECTED]
Sent: Friday, July 23, 2004 1:28 PM
Subject: RE: [Asterisk-Users] Priorizing of packets
Can HTB minimize latency better than CBQ?
Just curious,
Rich
Hmm I've had problems using HTB, it seems to make
There is always a risk of fraud says Jeffery Williams, CIO of Broadvox.
Our responsibility is to mitigate that risk as much as reasonably possible.
Revealing information such as the SIP credentials is, in my opinion, an
excessive risk.
This is the kind of closed-minded crap I would expect from
Hello
All,
I rebuilt my machine
adn there has been about 2 weeks time since my original CVS checkout. I
have seen teh changes for features.conf so that does nto worry me.
Hwoever, after moving over my saved conf files, things are not really
running. Does anyone know aht happened to teh
Brent Franks wrote:
I have some reports from users that occasionally DTMF will stop working in
voicemail and they will have to exit the system to get it to work again.
The useragents are Polycom IP 500's and I am using dtmfmode=rfc2833, with
Ulaw codec. This is all on an internal switched 100mb
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