Hi all...
I have been shopping around and noticed that licensed music on hold music
can be a bit expensive if you want to assemble a variety of types. Does
anyone know of an inexpensive source?
Thanks...
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I pulled the CVS tree at 11:30PM (approximately) Eastern on March 11. This
problem does seem to be fixed. But there is a new issue. It now seems
that adding the prefix that solved the problems before now causes the
"488 Not Acceptable Media" error response:
856.137213 192.203.175.9 -> 2
Asterisk -rvvv
-- Starting simple switch on 'Zap/7-1'
-- Executing MusicOnHold("Zap/7-1", "") in new stack
-- Started music on hold, class 'default', on Zap/7-1
-- Stopped music on hold on Zap/7-1
== Spawn extension (home, 6, 1) exited non-zero on 'Zap/7-1'
-- Hungup 'Zap/7-1'
Dan, why are you using phonejack with ulaw codec? g723 (format=slinear
only) is working just perfect with phonejack and iconnect :)
Lubo
Dan Fernandez wrote:
I found similar problems.
With my phonejack I can make a call with ulaw with decent quality (I have a
64k line).
However, with Messenger
> **ASTERISK SIP PACKET
>
> XXX Need to handle Retransmitting XXX:
> REGISTER sip:166.60.255.41 SIP/2.0
> Via: SIP/2.0/UDP 68.86.118.111:5060;branch=12720924
> From: ;tag=08e71f4b
> To: ;tag=08e71f4b
> Contact:
> Call-ID: [EMAIL PROTECTED]
> CSeq: 113 REGISTER
> User-Agent: As
I have a test SIP account set up with WorldCom and I have been trying to
have Asterisk register to the WorldCom server with no luck. It appears
that the SIP headers are different coming from Asterisk. I have included
a packet capture from a successful login with a Windows Messenger client
for refer
Hello everyone,
I was working on implementing several changes to asterisk. I had described
the changes and their reasons in several emails before I started the job. I
have completed the modifications some weeks ago; the system was then tested
extensively and is now in use in a production environme
IM not sure this is the only cause of the "480 temporarily unavailable"
message from Iconnect. I noticed that the dialer from Iconnect put a 20
wait time between calls.
Gregg
On Tue, 2003-03-11 at 20:53, Jim Archer wrote:
> --On Tuesday, March 11, 2003 5:31 PM -0800 John Todd <[EMAIL PROTECTED]>
Hi All...
I am using Asterisk on Debian with a single FXO card. I find that when I
dial into it it sounds very soft. I also noticed that when I record VM
greetings (I use the USB device for FXS) they are very soft.
I saw the rxgain and txgain. Can some one tell me how these are used? I
hav
Actually I think it was an issue with incrementing the sequence number on
the bye andshould be fixed now. RTCP is irrelevant in SIP signalling.
Mark
On Tue, 11 Mar 2003 [EMAIL PROTECTED] wrote:
> > 1 - From watching the udp fly by, it seems that iconnect does not know
> > when we hang up. For
--On Tuesday, March 11, 2003 5:31 PM -0800 John Todd <[EMAIL PROTECTED]>
wrote:
Because Asterisk doesn't implement RTCP.
That should have nothing to do with it, right? If a SIP "BYE" message
gets sent to the remote end by Asterisk, the RTP connection should get
shut down.Or am I missing some
I shouldn't have spoken so quickly.
I just tested an ATA-186 to verify what I had said in the negative,
and I find that leaving the ConnectMode set to 0x00060400 (the
default from the factory on v2.15) doesn't seem to make a difference
- my ATA behind a NAT worked just fine. Go figure.
JT
Y
> 1 - From watching the udp fly by, it seems that iconnect does not know
when we hang up. For example, if I call a voice mail and it starts
giving me its speal and I hang up, iconnect stays connected until the VM
hangs up at its end.
Because Asterisk doesn't implement RTCP.
That should have no
You may have read the earlier posts on it, but to make sure: you did
set ConnectMode to 0x00460400 right? Also, you have "nat=1" in the
settings for your ATA-186 in sip.conf? Both are required for correct
functionality as far as I've seen.
(Note: I've got an ATA-186 on a public IP address con
> 1 - From watching the udp fly by, it seems that iconnect does not know
> when we hang up. For example, if I call a voice mail and it starts
> giving me its speal and I hang up, iconnect stays connected until the VM
> hangs up at its end.
Because Asterisk doesn't implement RTCP.
__
turn off the secret then, or tell the phone to call.
Mark
On Tue, 11 Mar 2003, Eric Wieling wrote:
> Let us know if you ever get it working. I also have an ArrayVox
> phone that I've never gotten working.
>
> On Tue, Mar 11, 2003 at 05:16:27PM -0500, Raymond McKay wrote:
> > > For some reason,
I found similar problems.
With my phonejack I can make a call with ulaw with decent quality (I have a
64k line).
However, with Messenger I hear a brief horrible noise and that´s it.
- Original Message -
From: "Jim Archer" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, March 1
Jim,
I am seeing the same hangup problem.
The only client I am using with iconnect is their windows dialer. It
seems to work well.
Gregg
On Tue, 2003-03-11 at 18:17, Jim Archer wrote:
> Ok! When I use the prefix and I allow gsm it does work! And the
> quality is fine.
>
> There are two
I was able to install the rpms but when I run astman I get a segfault after
I try to login (independently of the user I use)
Yesterday I saw another posting regarding a segfault with astman.
Any suggestions?
- Original Message -
From: "Michiel Betel" <[EMAIL PROTECTED]>
To: <[EMAIL PROTE
Ok! When I use the prefix and I allow gsm it does work! And the
quality is fine.
There are two problems we're having now.
1 - From watching the udp fly by, it seems that iconnect does not know when
we hang up. For example, if I call a voice mail and it starts giving me
its speal and I
Hi everyone,
Has anyone else noticed that in the recent CVS builds that during
asterisk startup, the function build_peer now complains about the entry
"type=peer" or "type=friend" saying it will be ignored.
Looks like someone added a catchall else statement to build_peer, but
never added a case
Let us know if you ever get it working. I also have an ArrayVox
phone that I've never gotten working.
On Tue, Mar 11, 2003 at 05:16:27PM -0500, Raymond McKay wrote:
> > For some reason, this 407 Proxy Authentication Required seems to be
> > getting in the way... Any ideas? The UID and PW are fine
After sending my most recent message, I realized that I did not word it
clearly. In case of any confusion as to my configuration, my Asterisk
machine has a public IP, and the ATA186 is behind a NAT firewall. My
apologies for any confusion.
--
Matthew Farley <[EMAIL PROTECTED]>
__
> For some reason, this 407 Proxy Authentication Required seems to be
> getting in the way... Any ideas? The UID and PW are fine in the 186 (it
> works great when it isn't behind NAT).
>
I'd like to add that I have seen this same problem using Arrayvox Voxphones
(SIP phones also) both behind and n
yes, it's used by the ATA186 and others to try to discover their NAT
address.
Mark
On Tue, 11 Mar 2003, T Aksoy wrote:
> Hi Mark,
>
> Not familiar with "received=". What does it do? Has it got any application
> within the nat domain?
>
> Thanks
> Tan
>
>
> - Original Message -
> From: "M
I'd like to thank you all for the work that has been done on NAT
SIP support in Asterisk recently. This is truly the sort of project that
proves the strengths of open source development. I noticed that there
were a few people who had NAT SIP working (ATA 186 inside NAT, Asterisk
with public IP
It's received in a SIP header.
regards
Martin
On Tue, 11 Mar 2003, T Aksoy wrote:
> Hi Mark,
>
> Not familiar with "received=". What does it do? Has it got any application
> within the nat domain?
>
> Thanks
> Tan
>
>
> - Original Message -
> From: "Mark Spencer" <[EMAIL PROTECTED]>
> To
Hi Mark,
Not familiar with "received=". What does it do? Has it got any application
within the nat domain?
Thanks
Tan
- Original Message -
From: "Mark Spencer" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, March 11, 2003 8:25 PM
Subject: [Asterisk-Users] FIX: iconnect + del
>From now on (taking about asterisk's CVS) you can access environmental
variables using ${ENV(VARENV)}
regards
Martin
On Tue, 11 Mar 2003, Rattana BIV wrote:
> I try to detect if an user who use Netmeeting is connected or not.
> I think in order to do that, Netmeeting-user open a web page (in PH
On Tue, 2003-03-11 at 12:25, Mark Spencer wrote:
> Okay fellas, thanks to Ravi Sakaria's keen observations, we finally found
> what broke. When we added support for ;received=, it broke the ability to
> connect to iconnect. CVS Asterisk now only sends the ;received= if nat is
> turned on on the c
That shows true belief in your product (and supplier of crystals!) I'll try
it both ways, but I guess that, lacking a tone generator, I won't notice the
difference..
Thanks!
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Mark Spencer
Sent: dinsdag 11 maa
Okay fellas, thanks to Ravi Sakaria's keen observations, we finally found
what broke. When we added support for ;received=, it broke the ability to
connect to iconnect. CVS Asterisk now only sends the ;received= if nat is
turned on on the connection. Please try it out and get back to me.
Mark
I haven't play around enough to know whether or not the prefix is a
toggle. I will do some experimenting and let you know. Right now I am
prefixing all my calls with .
My experience is that when the carrier's format is G723.1, you can't
hear the incoming voice. When it is in G711 you can.
> Hi All...
Hi Jim
>
> Can an X100P be used as an FXS?
>
no.
> Thanks...
>
regards
kapejod
--
Klaus-Peter Junghanns
CEO,CTO
Junghanns.NET GmbH
Breite Strasse 13 - 12167 Berlin - Germany
fon:+49 30 79705390
fax:+49 30 79705391
iaxtel: 1-700-157-8753
email: [EMAIL PROTECTED]
__
> Thanks! I'll set the dipswitch on the CAC to provide sync, currently the T1
> is free running, but its only been up for an hour or 3 so no slips yet.
Actually you probably want to supply the CAC with sync.
> Will excessive slips generate a yellow alarm? Then I'll know what to
> watch...
Slips
NO
Jim Archer wrote:
Hi All...
Can an X100P be used as an FXS?
Thanks...
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On Tue, 2003-03-11 at 14:01, Jim Archer wrote:
> Hi All...
>
> Can an X100P be used as an FXS?
No, it is a FXO only card.
--
Steven Critchfield <[EMAIL PROTECTED]>
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Thanks! I'll set the dipswitch on the CAC to provide sync, currently the T1
is free running, but its only been up for an hour or 3 so no slips yet.
Will excessive slips generate a yellow alarm? Then I'll know what to
watch...
Michiel Betel
-Original Message-
From: [EMAIL PROTECTED]
[mail
Hi All...
Can an X100P be used as an FXS?
Thanks...
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Sure, the Digium T1 can be left in free-run. The place to set up loop
(or line) timing is in the CAC channel bank. That way there wouldn't be
slips.
Don Pobanz
On Tuesday, March 11, 2003 1:33 PM, Mark Spencer
[SMTP:[EMAIL PROTECTED] wrote:
> > Question is how I should set the sync timing in za
> Question is how I should set the sync timing in zaptel.conf though, The E1
> gets sync from the network, but can an E1 slave to that? (E1 -T1 timing is
> quite different) or should I let the T1 sync to the CAC?
You will leave the E1 syncing from the network and the T1 in free-run.
You will defin
Hi Greg and thanks very much...
A few questions...
First, regarding the prefix, it seemed that this acts as a toggle,
switching from the one codec to the other. But how do I set which me
system uses by default? Or does iconnect always use the high bandwidth one
by default (such that the
On Wed, 5 Mar 2003 21:33:08 +0100
"Michiel Betel" <[EMAIL PROTECTED]> wrote:
> I don't know (haven't tried myself) but Kostya V. Ivanov's 'R' patch to the
> dial application (december 2002) might be of help for you. Check the
> archives for Barge (Intrusion) Capabilities. It might be some manual w
On Tue, 11 Mar 2003, Tilghman Lesher wrote:
> On Tuesday 11 March 2003 08:44, Steven Critchfield wrote:
> > On Tue, 2003-03-11 at 02:29, Rattana BIV wrote:
> > > I try to detect if an user who use Netmeeting is connected
> > > or not. I think in order to do that, Netmeeting-user open a
> > > we
Title: Message
Today we finally set
up our testbox, an digium E1 connected to the outside world and a digium T1
connected to a CAC-accesbank-1. After some fiddling and finding out the T1 cable was
faulty we got everything working. Great when the analog phone starts
ringing!!
Question is ho
On Sat, 2003-03-08 at 15:51, William X Walsh wrote:
> Another phone (untested, they just posted to the FWD list earlier)
>
> http://www.xten.com/products.php?menu=Products
>
> Again, no pricing, but I've emailed them for more info.
I joined their beta test program, and started testing their beta
Jim,
I changed my extensions entry for iconnect to:
exten => _1XX,1,Dial,SIP/[EMAIL PROTECTED]
and my calls work fine with ulaw. I am calling from a linejack card
with format=ulaw and SIP with allow=ulaw.
Gregg
On Mon, 2003-03-10 at 23:01, Jim Archer wrote:
> --On Monday, March 10, 200
The most common R2 version, MFC R2. How are working in this project ?
Regards,
Peter
On Tue, 11 Mar 2003 21:43:46 +0800
Steve Underwood <[EMAIL PROTECTED]> wrote:
> Claudio Aznar wrote:
>
> >Hello,
> >
> > I'm testing the E400P with PRI signaling and work fine, but I want to test it
> > wi
On Tuesday 11 March 2003 08:44, Steven Critchfield wrote:
> On Tue, 2003-03-11 at 02:29, Rattana BIV wrote:
> > I try to detect if an user who use Netmeeting is connected
> > or not. I think in order to do that, Netmeeting-user open a
> > web page (in PHP) et press the button Connect or Disconnect
Newt is no longer included in SuSE 8.1, I tried installing the 7.2 newt
packages but they don't work correctly, finally I installed the 7.2 source
rpms and rebuilt them for 8.1, that works
Michiel
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ben Klang
On Mon, 2003-03-10 at 05:13, William X Walsh wrote:
> I have had no problems with call quality, but right now inbound calling
> from them is having an issue with completing the call through to
> asterisk.
Just to make sure the other d3/iconnecthere users on the list know, this
problem has been fi
Hi there,
just wanted to let you know that i will be wandering around on
CeBIT this Friday the 14th. If anybody wants to meet and bug me with
some capi, isdn or general asterisk stuff contact me off list.
If not i will sit silently in the corner and wait for my train
back to Berlin ;-)
regar
On Tue, 2003-03-11 at 05:41, James Sizemore wrote:
> I got the time to go back and investigate the problem.
> You may want to comment out messages in logger.conf
> by default, or have a comment in the README to add
> /var/log/asterisk/messages to the log rotation.
>
> I found the max file size f
On Tue, 2003-03-11 at 02:29, Rattana BIV wrote:
> I try to detect if an user who use Netmeeting is connected or not.
> I think in order to do that, Netmeeting-user open a web page (in PHP) et
> press the button Connect or Disconnect and the PHP set the Environnement
> variable which will be proceed
Nir Simionovich wrote:
Hi Steve,
Hmmm lets see, NOT !
In Israel the connection of the E1 at the wall socket is not an RJ45, but
a British Connector.
I'm fully aware of the fact that the connection of E1 on the Router/Server
side are 1,2 and 4,5 as pair for TX and RX. Steve, my question ma
Hallo all
I’m had a look in the archives but
could not find any specific messages pertaining to configuring Asterisk to work
with a isdn external TA that is CAPI 2.0 compliant
Is this possible, if not could you
please refer me to a message on how to configure asterisk to work with an
i
Claudio Aznar wrote:
Hello,
I'm testing the E400P with PRI signaling and work fine, but I want to test it
with R2 signaling.
My question are:
1. Can the E400P work with R2 signaling ?
2. If the firs question is yes, HOWTO?
Thank in advance
Peter
Is it available? N
I got the time to go back and investigate the problem.
You may want to comment out messages in logger.conf
by default, or have a comment in the README to add
/var/log/asterisk/messages to the log rotation.
I found the max file size for ext3 file system. LOL
-rw-r--r--1 root root 2147
> As far as I remember the 4 port cards will be selecteable FXS/FXO.
> But better to verify with mark or greg :)
>
> > > Similar, but not the same. You can try to use the USB adapter, or if you
> > > wait shortly you will be able to get the 4 port FXS card.
> >
> > Is there any rough estimate on
As far as I remember the 4 port cards will be selecteable FXS/FXO. But better
to verify with mark or greg :)
On Tuesday 11 March 2003 09:49, Jim Archer shaped the electrons to say:
> --On Monday, March 10, 2003 11:43 PM -0600 Steven Critchfield
>
> <[EMAIL PROTECTED]> wrote:
> > Similar, but not
Hi all...
I was wondering if anyone has upgraded from an X100P FXO and now would like
to turn it into cash?
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--On Monday, March 10, 2003 11:43 PM -0600 Steven Critchfield
<[EMAIL PROTECTED]> wrote:
Similar, but not the same. You can try to use the USB adapter, or if you
wait shortly you will be able to get the 4 port FXS card.
Is there any rough estimate on when this will be available?
Also, are there
I try to detect if an user who use Netmeeting is connected or not.
I think in order to do that, Netmeeting-user open a web page (in PHP) et
press the button Connect or Disconnect and the PHP set the Environnement
variable which will be proceeded in extension.conf
So i need Environnement Variable,
The newt development package can be installed via yast2.
run 'kdesu yast2' or 'su -c yast2', find the Install Packages option,
and search for newt.
Hope that helps,
-BAK
On Mon, 2003-03-10 at 16:26, Dan Fernandez wrote:
>
> Can astman be compiled without newt? I have Suse 8.1 and it doesn´t
>
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