-Original Message-
From: Benjamin on Asterisk Mailing Lists
[mailto:[EMAIL PROTECTED]
Sent: September 7, 2004 10:53 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Maximum tollerable lag/jitter
for IAX2 w/o jitterbuffer enabled?
On
On Wed, 8 Sep 2004 01:04:53 +0100, asterisk-users
[EMAIL PROTECTED] wrote:
I have successfully used the packaged version of * on the Mac for some
time, but decided that I would recompile one of the more recent builds
so that my PC and Mac were in sync.
As suggested, I installed the XCode
Renato Mintz wrote:
Hi Folks,
I've been looking around and found some references of some Caller ID
patches (Mantis bug#9) for X100P and TDM400 for Netherlands, Sweden
and UK. It's been quite hard to understand what has finally been
incorporated to the distribution (if anything) or which
I had that problem when I was running asterisk on
my linux box before it went down
so you aren't the only one having that
problem
- Original Message -
From:
Marty
Mastera
To: Asterisk Users Mailing List -
Non-Commercial Discussion
Sent: Tuesday, September 07, 2004
hello I am fitteling with the
astwind-installer-0.1.1.exe asterisk for windows and am having trouble getting
the thing to connect to the meers to download the updates and stuff. I
looked at the wiki and set up networking and stuff with no success, has any one
got this thing to work
I was looking at the
sample "follow me" config (http://www.voip-info.org/wiki-Asterisk+Tips+follow+me) which uses a dial modifier 'c' to enable Answer
confirmation - "If the letter c follows, then "Answer Confirmation" is
requested, in which the call is not considered answered until the
Hi Marty
I think this is a valuable add-on to my feature request to play audio
until # is pressed - see
http://bugs.digium.com/bug_view_page.php?bug_id=0002356
At the moment just hearing silence is not very helpful to users.
Perhaps we can extend this feature to
a) provide this support on
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
Tim Robinson
Sent: Wednesday, September 08, 2004 1:39 AM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [Asterisk-Users] Answer confirmation on non-Zap channels?
Hi
In France we just have a very short ring before the CID spill.
The CID spill is encoded in V23 instead of Bell 202
If you want the X100P to decode CID try this in fskmodem.c :
#define FLIST {1400,1800,1300,2100}
and maybe (i don't remember !) callerid.c :
#define CALLERID_SPACE 2100.0 // CCITT
Hi everyone!
I have a problem... We have received a couple of phone numbers for voip
from a local voip-provider. The work fine directly with a Cisco 7960,
but so far I've not been able yet to integrate them into Asterisk.
I've tried:
/etc/asterisk/extensions.conf
*
[ip-incoming]
I made the observation that I'm able to make a call with my SIP client (kphone) even
when I'm not
registered/authenticated.
Of course, when I'm not registered at asterisk, people can't call me, but it's still a
huge security hole,
that unregistered Clients can make calls.
Is there a way to
On Wed, 08 Sep 2004 10:08:00 +0200, Evert Meulie [EMAIL PROTECTED] wrote:
I have a problem... We have received a couple of phone numbers for voip
from a local voip-provider. The work fine directly with a Cisco 7960,
but so far I've not been able yet to integrate them into Asterisk.
I've
Have you tried running mpg123 from the command line?
I found that it was failing to load the mp3 cos it couldnt open /dev/dsp.
If this is the case, then I found the following worked, although it is a
little OTT I guess...
If you have a soundcard in the machine try insmod'ing a driver for it and
On Wed, 8 Sep 2004 10:31:44 +0200, Henry Jensen [EMAIL PROTECTED] wrote:
I made the observation that I'm able to make a call with my SIP client (kphone)
even when I'm
not registered/authenticated.
Of course, when I'm not registered at asterisk, people can't call me, but it's still
a huge
Variables DIALSTATUS: added to CVS head in june/july 2004!
What is your CVS version?
Areski
On Wed, 2004-09-08 at 03:44, Doug Harris wrote:
Hi,
I have set-up astcc with outgoing sip channel. Call processing works
fine but after the call tables, CDR and Cards does not get updated. At
the
HI!
Have a weird problem with X-lite Meetme.
When X-Lite user are join to conference room NOT first one, than
X-Lite user do not hear anything. This problem gone when X-Lite user get
into conference room first (when nobody there).
sip.conf
[104]
context=VoIP-only
type=friend
username=104
I'm wondering if you are confusing two ideas. It has to be possible for
anyone to be able to call you just like they can on an ordinary POTS line.
Registration is for those who need to appear in some sense internal to the
PBX. Using dialplan contexts you can offer very different functionality to
Hi all,
quick question...i am using autocreatepeer to get asterisk to work with SER
without having to specify each UA in sip.conf and in ser separately.
2 questions:
1. obviously this is not very secure because anyone can bypass the SER
and register themselves as a peer with the asterisk. assuming
I'd thought I'd been through the whole Zapata Telephony Site. Could you
e-mail back and point to the specific links you had in mind?
Start with
http://www.zapatatelephony.org/philos.html
and dive into
http://www.zapatatelephony.org/project.html
and then into
Hi!
Sample configuration or other documentation from the provider? Hmm,
haven't received any! :-/
all I got was username password...
Is there a way (perhaps with sipsak?) to determine what kind of
server/system they are running?
If their system is not IAX-compatible, what are my options then
And the same problem with Grandstream HandyTone-286 as well
On Wed, 2004-09-08 at 11:43, Vladyslav wrote:
HI!
Have a weird problem with X-lite Meetme.
When X-Lite user are join to conference room NOT first one, than
X-Lite user do not hear anything. This problem gone when X-Lite user get
Hello!
Since i was not able to compile chan_capi 3.5 on either Fedora2, Debian
stable/testing/unstable i decided to use the normal sources, and then
patch chan_capi with the debian patch. Now i can compile chan_capi woth no
errors.
When i start asterisk on Fedora2 (2.6.5-1.358smp), i get this:
.
On Tue, 7 Sep 2004, Chris Shaw wrote:
All calls are running as GSM, even though g.729 is also an 'allowed' codec
(w/5 licenses installed). During an average call 'iax2 show channels'
provides:
Peer UsernameID (Lo/Rem) Seq (Tx/Rx) Lag Jitter
JitBuf Format
On Tue, 7 Sep 2004, Kris Boutilier wrote:
Reproducing it is part of the problem. I've been getting user reports on and
off for some time but I can't find anything out of the ordinary - initially
this was looking like a Voicemail bug as many people were getting cut off
while leaving
Hi there !
I searched the whole web to find some helping information about H323
Control Protocol, but there is no way to find that information.
We compiled and installed asterisk_0.9.0 + pwlib 1.5.2 + openh323_1.12.2
+ 'asterisk-oh323_1.5 channel driver + wrapper' and configured the
dialplan for
On Tue, 7 Sep 2004, Kris Boutilier wrote:
The only thing that springs to mind, if it's all UDP driven, is a lack of
retry handler for the UDP handoff acknowledgement? I'm averaging about a
0.5% collision rate on this network (half-duplex 10Base-T)...
My IAX connections soldier on over
Hi,
I have on my dual opteron (64 bit mode, linux) the problem, that
sometimes read errors (unknown error 500) occur.
This was already discussed on some asterisk list, and the solution
seems to be to put the digium card on the highest interrupt level.
Unfortunately I don't know howto. Applying an
On Wed, 2004-09-08 at 13:43, HengWee Chin wrote:
Hi all,
I have the following setup
PSTN - ASTERISK - IVR (using dialogic card)
1) Caller id information is presented to asterisk during the first and
second ring.
2) Hence, Asterisk waits for 2 rings before pickup the call and
Hi
i've an hfc-s card with last bristuff installed
at cli i'm receiving:
Sep 8 12:35:20 WARNING[1109552048]: chan_zap.c:6902 zt_pri_error: PRI: !! Got a UA,
but i'm in state 1
received TEI check request for TEI = 77
what is causing them?
10x
Maurizio
Good day all
I'm just wondering for interest sake
I have a box,hostname=myname.co.za,running sendmail
If I send mail to [EMAIL PROTECTED] it try to deliver to the box,witch
does not have the mail box.How do I tell sendmail that it should send mail to
myname.co.za's mailserver.
I know how easy it
get a book on DNS andlookup MX records or look on yolinux for a tutorial
Altus Snyman wrote:
Good day all
I'm just wondering for interest sake
I have a box,hostname=myname.co.za,running sendmail
If I send mail to [EMAIL PROTECTED] it try to deliver to the box,witch
does not have the mail box.How
Hi!
It turns out my provider uses the Micronet SIP server. Any possibilies
to let this one interface with Asterisk?
Regards,
Evert
Evert Meulie wrote:
Hi everyone!
I have a problem... We have received a couple of phone numbers for
voip from a local voip-provider. The work fine directly with a
I'm trying to set up xinetd to run an asterisk console on a tcp port.
So far I've added a file in /etc/xinetd.d/ like:
service actl
{
disable = no
socket_type = stream
protocol= tcp
port
hi all
I'm trying to setup call divertion with the standard
*21*numbertodivertto#
etc
but...
When I dial such a number from a SIP client, it generally works quite
badly
most of the ones I've tried can handle *, but none, or at least few,
can handle #
Is this a SIP protocol weakness, or what is
On Wed, 8 Sep 2004, Roy Sigurd Karlsbakk wrote:
hi all
I'm trying to setup call divertion with the standard
*21*numbertodivertto#
etc
but...
When I dial such a number from a SIP client, it generally works quite
badly
most of the ones I've tried can handle *, but none, or at
On Sep 7, 2004, at 7:15 PM, Chris wrote:
Asterisk never ever uses TCP for IAX or IAX2. It's ALWAYS UDP. I
don't
believe Asterisk supports SIP over TCP either. Heck, the manager port
is the only thing that uses TCP that I know of with Asterisk.
Hmmm I wonder why I had the impression that it
On Sep 7, 2004, at 4:43 PM, Chris Shaw wrote:
- Original Message -
From: Andrew Thompson [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List - Non-Commercial Discussion'
[EMAIL PROTECTED]
Sent: Tuesday, September 07, 2004 4:39 PM
Subject: RE: [Asterisk-Users] MeetMe without ZAP?
Matthew
hi,
i had asterisk and gnugk running on fedora core 2. it worked quite well. then, i needed
to change to red hat 9, and i'm experiencing troubles with h.323 :-( making a call from
a h.323 phone (innovaphone) does not work, and dial-in also doesn't. below
is an excerpt of what happens, when i try
Hello,
I would like to know, if it is possible to accept DTMF signals from a
caller while he is in a queue.
I would like to accomplish something like this:
1) The caller is in the queue.
2) The caller dials 123.
3) The caller is sent to extension 123.
just for your information:
When the caller
filed as 0002399
On 8. sep. 2004, at 14.25, [EMAIL PROTECTED] wrote:
On Wed, 8 Sep 2004, Roy Sigurd Karlsbakk wrote:
hi all
I'm trying to setup call divertion with the standard
*21*numbertodivertto#
etc
but...
When I dial such a number from a SIP client, it generally works quite
badly
most of the
Hello,
On Wed, 8 Sep 2004 12:54:50 +0100 (BST), Mark Turner [EMAIL PROTECTED] wrote:
I'm trying to set up xinetd to run an asterisk console on a tcp port.
So far I've added a file in /etc/xinetd.d/ like:
snip
After adding actl to /etc/services and restarting xinetd it reports
one new
It specifically says here: http://www.voip-info.org/wiki-Asterisk+timer
that zaprtc cannot be used with an SMP machine. What is this timer used
for that it is so important?
Matthew
- Original Message -
From: Andrew Thompson [EMAIL PROTECTED]
To: 'Asterisk Users Mailing List -
Hi,
I tried to play the standard demo-echotest file !.
It works when i use an ip-phone (like x-lite or kphone), but as far as i use an PSTN Gateway (from an VOIP Provider) to call my phone - i get the following error:
Sep 8 14:58:33 NOTICE[-182461520]: channel.c:1691 ast_set_write_format:
On Tuesday 07 September 2004 20:55, Kris Boutilier wrote:
The arrangement right now has:
PSTN Trunks Stations - Nortel Norstar#1 -CT1- Asterisk#1 -IAX2-
Asterisk#2 -CT1- Nortel Nortstar#2 - Stations
The Asterisk boxes provide Voicemail to their sites Norstars and intersite
calls over IAX.
Hello,
On Wed, 8 Sep 2004 13:10:48 +0200, Altus Snyman [EMAIL PROTECTED] wrote:
Good day all
I'm just wondering for interest sake
I have a box,hostname=myname.co.za,running sendmail
If I send mail to [EMAIL PROTECTED] it try to deliver to the box,witch
does not have the mail box.How do I
Yes, I *am* using BROADVOICE, thanks for responses. Sure enough. If I dial in via my
Washington number (ipkall), I don't have the problem. Interesting. Well, BV has a very
good tech that seems to be very familiar with Asterisk. I'll see if he has any ideas
how to deal with the issue.
Sorry I
On Tuesday 07 September 2004 19:39, Chris Shaw wrote:
If you can reproduce it, this smells like a bug... IAX runs over TCP and
TCP doesn't just disconnect sockets unless it recieves a RESET or a
FINISHED or there's a timeout (usually like 5 minutes or more depending on
your TCP/IP stack).
After small review of the chan_sip.c you should turn on pedantic sipchecking
pedantic=yes in sip.conf [general]
bkw
-Original Message-
From: [EMAIL PROTECTED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Roy Sigurd Karlsbakk
Sent: Wednesday, September 08, 2004 8:16 AM
Hi,
I did a cvs update on 03 Sep.
How do I find out all available variables (to agi) in a particular code
version. I tried show agi get variable, but that wouldnt give me much
info.
Cheers
dh
-Original Message-
From: Areski [mailto:[EMAIL PROTECTED]
Sent: Wednesday, September 08,
Hi all,
I am an MTech student and currently working on a project on GSM air
interface. I am making use of Asterisk soft PBX. I am stuck at a point
regarding this. As far as I understood from the available Asterisk
documentation that Asterisk can easily plug into it the various programming
Hi,
I have just completed the deployment of a couple of Grandstream phones
(for internal IP use) and was wondering how much harder it would be to
deploy a Polycom IP 300 phone. The Grandstream was quite easy to deploy
and gives us good voice quality over DSL, however from some of the
previous
Hi all,
I just modified one of the startup scripts provided on the tarball to
fit on my SuSE 9.x system to start/stop Asterisk when the system boots
or goes down.
Maybe I'm overseeing the answer but could't find where to
post/(cvs)upload the changes I made...
TIA,
Martin
On Wed, 8 Sep 2004 18:58:11 +0530, Renu Rangnekar
[EMAIL PROTECTED] wrote:
As far as I understood from the available Asterisk
documentation that Asterisk can easily plug into it the various programming
interfaces and different codecs in it can seemlessly talk to one another.
Asterisk has a
Hi,
I recompiled asterisk today from CVS and Ive
been having a number of problems, Ive read the deadlock page on the wiki
and some of it sounds like that, however, the latest issue were having
it that sometimes Asterisk doesnt seem to know the PRI channel has
dropped, and assumes its
Hello Fabian,
Wednesday, September 8, 2004, 5:14:10 PM, you wrote:
FM I would like to know, if it is possible to accept DTMF signals from a
FM caller while he is in a queue.
A context may be specified, in which if the user types a SINGLE digit
extension while they are in the queue, they will
Hello Johannes,
Wednesday, September 8, 2004, 5:21:53 PM, you wrote:
JH Unable to find a path from GSM to G729A
Use Google. You'll need a license for G.729.
http://www.digium.com/index.php?menu=asterisk_g729
--
Best regards,
Olegmailto:[EMAIL PROTECTED]
hey * folk,
am trying to configure a WellGate 3504A FXS SIP ATA
(http://www.welltech.com.tw/products_ea01.htm) with asterisk. i've set
up two SIP clients in sip.conf as follows:
[general]
port = 5060 ; Port to bind to
bindaddr = 0.0.0.0 ; Address to bind SIP
Holger Schurig wrote:
I'd thought I'd been through the whole Zapata Telephony Site. Could you
e-mail back and point to the specific links you had in mind?
Start with
http://www.zapatatelephony.org/philos.html
and dive into
http://www.zapatatelephony.org/project.html
and then into
I would be interested in the script. Did you do zaptel drivers too?
On Wed, 2004-09-08 at 10:41, Martin Mielke wrote:
Hi all,
I just modified one of the startup scripts provided on the tarball to
fit on my SuSE 9.x system to start/stop Asterisk when the system boots
or goes down.
Maybe
Good time of day all!
1)
I am trying to use as5300 and asterisk. As5300 sends calls to me. I
get the following in
* console:
-- IAX2/magrathea/6 is making progress passing it to OH323/R27464
Sep 8 10:57:59 NOTICE[1140046640]: chan_oh323.c:1159 oh323_indicate:
Ignoring PROGRESS indication.
Just a comment here.
I built an * pbx on a Celeron 1.4ghz machine. Got all the dialplan and such
working , then built a new server with an AMD 2.4g processor with a 500mhz
front side buss. With the same Digium TDM cards and all analog incoming and
outgoing.
The celeron was not ringing out until
Tony Nichols wrote:
I would be interested in the script.
OK. I'll send it off the list...
Did you do zaptel drivers too?
Nope ;)
Martin
___
Asterisk-Users mailing list
[EMAIL PROTECTED]
http://lists.digium.com/mailman/listinfo/asterisk-users
To
-Original Message---
Yes, I *am* using BROADVOICE, thanks for responses. Sure enough. If I dial
in via my Washington number (ipkall), I don't have the problem. Interesting.
Well, BV has a very good tech that seems to be very familiar with
Ben,
I ran into a similar issue on the 8/31 cvs, except it was backwards.
Outbound calls would report a busy on the channel selected, yet a few
minutes later the channel would be used for an inbound call. I had to
revert back to my previous checkout from 8/16 to resolve the issue. The
problems
Try, in the 53 (depends on the SW version u're using
voice call send-alert
Also if you're using PRI trunks you can use, in the Serial interface,.
isdn send-alerting
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Maxim
Litnitsky
Sent: Wednesday,
Update to latest CVS and/or put context=INVALID in [general] in sip.conf
and in each peer/user/friend entry put in a correct context= line.
On Wed, 2004-09-08 at 03:31, Henry Jensen wrote:
I made the observation that I'm able to make a call with my SIP client (kphone)
even when I'm not
I think this issues stems from the (in my case) wct4xxp driver. When
updating libpri, I also updated zaptel, however, I'm unsure if I
installed it correctly (i.e. updated to the newly compiled version).
After stopping asterisk, doing rmmod wct4xxp, make install on zaptel and
then restarting
Huddleston, Robert wrote:
I'd like a startup script for redhat... should be just some small changes..
do you have one?
It's already there... :-)
Take a look at .../asterisk_v1_0_stable/contrib/init.d to find a file
called rc.redhat.asterisk. This one should do the trick... ;)
HTH,
Martin
On 08/09/2004 20:29 Dinesh Nair said the following:
am trying to configure a WellGate 3504A FXS SIP ATA
(http://www.welltech.com.tw/products_ea01.htm) with asterisk. i've set
up two SIP clients in sip.conf as follows:
i forgot to mention that i'm running asterisk 0.90.0 on freebsd 4.10 and
the
Il 05:41, mercoledì 08 settembre 2004, Steve Maroney ha scritto:
On my default asterisk installation, *80 didn't work until I modified the
source to move call pickup to *9. I wasn't sure what I was doing but *80
works now. Except I thought *80 would play some voice prompts that gave
the option
I have searched through the wiki, but I was unable to find the
information that I desire. I am trying to implement the Directory
command within my Asterisk configuration. What information is passed
back when a name is successfully found? Since Asterisk is being used as
an automated attendant
Hi,
I just installed OH323 Plugin and im now tryin to
make
simple Configuration to connect Openphone and Xlite
to
my Asterisk-Server.
All works fine, i just wanna know if there's a
better way to do it? Is there anything wrong with my
Config?
OH323.conf
[general]
I'm using Asterisk to read voicemail users out of a SQL database. I am
assigning users real phone numbers as their voicemail box. The problem is
that if I re-assign a phone number (say, 972-245-0001), the new user is
stuck with the old user's greeting and saved messages. What is the best way
to
All,
i am new to asterisk, and I have been searching through the list and
docs for examples on howto accomplish this, but i havent had much luck.
1. have asterisk answer when an unregistered cisco gateway send its a
SIP call -- DONE. (using the demo samples i successfully get into
the demo)
Hi !
I have a E100P and I would like to receive incoming calls on dedicated
channels only.
Is it possible to answer an incoming call request on channel 2-30 from
the Telco with something like 'busy, use channel 1 instead' ?
If this is possible, how could it be implemented / configured ?
Br / Jan
Thanks for the tips ...
Like you said, dealing with carrier is not going to get me anywhere.
The only thing GT recommended was grounding the server chasis :P
I turned the echo cancellation with the same parameters you used and
It doesnt even make a difference. I dug further into the
Hi,
I get an Intertex IX66 and Im trying to connect my *
behind this SIP router. I can register my Polycom phones on * but the sound on
the phones is just one way.
Someone fight with the same problem with this router?
Any suggestions are really appreciated.
Best regards,
Chris
Oleg A. Arkhangelsky [EMAIL PROTECTED] writes:
A context may be specified, in which if the user types a SINGLE
digit extension while they are in the queue, they will be taken out
of the queue and sent to that extension in this context.
See queues.conf.
Wow, thanks a lot Oleg. I overlooked
Hey,
I've checked all over and can't find what I need to
know, so I'm posting here. I want to use Asterisk
with my Primus VoIP service but it seems I need a
username and password to authenticate with at Primus.
Has anyone had any experience with this? How did you
get it? Is it stored
On Wednesday 08 September 2004 13:39, [EMAIL PROTECTED] wrote:
None of these were listed at all in the Makefile, so I added them
And tried a recompile. Still a bad echo. It is like the echo
cancellation
Is not even working. Is there a way to verify its active or not?
It's not in the makefile,
I have asked before and got no answers - I am still not clear as to why
there is not an MGCP client as part of asterisk - is it a technical
reason, no one else wants it, other ?
It is my understanding primus is using mgcp, and therefore is unable to
directly interface with asterisk, password or
Hello,
I recently upgraded to Sip 1.3.1 and noticed that the Reject Button is
no longer appearent on the screen when a second incoming call comes in
unless I press the hold button on the first call.
Does anyone have a work around for this to reject a call while
continuing to talk to the first
At the moment we're not - the email notification from Comedian Mail has
been mostly sufficient. I do however have some Dialogic D/42-NS PBX
emulation cards and the plan is to use them to set and unset the MWI lamps
based on events pushed out of Asterisk.
They may be obsolete hardware but they
Well I wouldn't look at my question as a problem. I thought I would get
more functionality out of *80 after seeing other sound files in my sounds
directory. Those voice prompts look useful so maybe I am missing
something here.
Thank you,
Steve Maroney
On Wed, 8 Sep 2004, Diego Ercolani wrote:
Yeah it looks to be the same setup as mine I am going to try out
Mark3 and the Aggressive Suppresor as well.
Paul Seniuk
-Original Message-
From: akohlsmith-asterisk [mailto:[EMAIL PROTECTED]
Sent: September 8, 2004 12:34 PM
To: asterisk-users
Subject: Re: [Asterisk-Users]
Collecting debug today - got 8mb so far. Just turned up IAX2 debug so
expecting it to balloon.
There is no NAT involved, nor stateful firewalling etc. - this is a flat
10.0.0.0/24 subnet with one 10base-t hub and two 10base-t cable modems
(operating peer to peer) in between the endpoints. Bitrate
Jon,
Does Primus actually use MGCP though? I've heard mix
results (though keep in mind I only became interested
in all of this earlier today, so I know very little).
I checked the specs on my dlink and it says it's SIPs
with no mention of MGCP. However everywhere else says
Primus is not SIPs.
Jon,
Does Primus actually use MGCP though? I've heard mix
results (though keep in mind I only became interested
in all of this earlier today, so I know very little).
I checked the specs on my dlink and it says it's SIPs
with no mention of MGCP. However everywhere else says
Primus is not
On Wednesday 08 September 2004 14:46, [EMAIL PROTECTED] wrote:
Yeah it looks to be the same setup as mine I am going to try out
Mark3 and the Aggressive Suppresor as well.
I'm using Mark2 and *no* agressive supressor (which surprised me, I thought I
had it in there)
-A.
[EMAIL PROTECTED] wrote:
Jon,
Does Primus actually use MGCP though? I've heard mix
results (though keep in mind I only became interested
in all of this earlier today, so I know very little).
I checked the specs on my dlink and it says it's SIPs
with no mention of MGCP. However everywhere
Jon,
Hmm I didn't know about the versions thing. I'll have
to get the exact model number off the device when i
get home.
I never set up any forwarding at all for it though. I
simply plugged it into my switch and everything was up
and running within a few seconds. Not sure if that's
a good
Geoff,
How frequent are your dropped calls? For a while all
my calls would go silent but I realized it was after
exactly 60 minutes. It's since been increased to 180.
Not sure if this is what you were experiencing.
Are there any providers in Canada that offer a similar
service to Primus that
On Wed, 8 Sep 2004, Kris Boutilier wrote:
I'm leaning in the direction of an
IAX2 ACK packet being dropped off the network - I've noticed about 0.5%
collisions on the wire (this being a half-duplex network) which seems to be
contributing to audible pops and clicks with the jitter buffer
I did a packet sniff and it is definitely MGCP.
I find that the quality hasn't been great. I am
looking at moving to
something different. It frequently drops calls, but I don't know if
it is the NAT device that does it. I would like to find
something that is a
little bit more Asterisk
We recently had a customer install that went horribly wrong. Serious
echo (pots lines into a cac cb) that, although * did a good job
getting rid of alot of it, could not get rid of it all. We tried
everything, every canceller, gain setting, etc... combination
possible to no avail.
Both the
I want the ability to setup DIDs in a variety of different remote locations
in Canada. There are various providers that have DIDs in major cities, but
none that focus on the smaller cities.
The question is how do I actually setup these DIDs?
Thanks,
Geoff
Sep 8 15:38:23 WARNING[180235]: chan_sip.c:595 __sip_xmit: sip_xmit of
0x8ec83b4 (len 434) to 147.135.0.129 returned -1: Bad file
descriptorSep 8 15:38:23 WARNING[180235]: chan_sip.c:595 __sip_xmit:
sip_xmit of 0x8ebb95c (len 434) to 147.135.8.128 returned -1: Bad file
descriptorSep 8
Greetings folks;
I have a bit of a conundrum, and I can't tell if Asterisk is doing
something daft, or whether I'm clean missing out why it's doing what it's
doing. So, I have a dialplan that looks a little like this:
[start]
include = dids
include = everythingelse
[dids]
On one of my 3 * servers I get this after 2 or 3 IAX2 calls
Apr 22 15:54:39 NOTICE[1150495040]: chan_iax2.c:1271 iax2_destroy:
Avoiding IAX
destroy deadlock
And as if that wasn't enough I get a never ending stream of this error
flying off the top of the screen. At which point I can no
Just recently installed a multitech mvp810 instead of a t100p and cac
adit channel bank.
Works perfectly, got rid of all echo issues that nothing else had been
able to (all the zap echo cancelers, mediatrix gateway, vegastream
gateway, etc etc...)
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