can just restart the machine but it is in a production environment
(soon to increase from a few to ~30 simultaneous calls) and it is nice to be
able to make changes and
cvs update installs without restarting.
Has anyone experienced this or am I just missing a step or going in the
wrong order?
John
> Did you ifdown the dynamic interfaces first ?
> Martin
I probably tried
/etc/rc.d/init.d/network stop
I was playing with this a few days ago so I don't remember all the
details... I'll collect a few more details.
John
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conf...
span=1,2,0,esf,b8zs
span=2,1,0,esf,b8zs
dynamic=eth,eth0/00:0A:5E:05:7E:89,24,0
> On Wed, 2003-12-17 at 10:36, john wrote:
> > Hi,
> >
> > I have just begun working with TDMoE running between 2 fiber nics the
> > dynamic span works great. In my main asterisk b
1) Is it possible to store the menu sounds in wav
...sure, just put your 8kHz 16 bit mono files named whatever.wav in
/var/lib/asterisk/sounds - asterisk will convert them to what is needed if
needed.
John
This e-mail was scanned and found clean by Monroe-Woodbury's Anti
that is what it was.
John
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of James H.
Cloos Jr.
Sent: Friday, January 02, 2004 1:03 AM
To: JR Richardson
Cc: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] * crash when forward voicemail --Nicolas
Gudino
&
o run directly on
the wav files? Now-a-days hard drives are so big, why use compression at all
(at least for local files)?
John Harragin
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[EM
or a while & I
don't want to jump into echotraining without a way to quickly enable what
has been working for me.
How do I enable this mode now?
How are people liking conversations with the echotraining enabled on both
ends of connections like...
remote* <> iax <> * pstn what
e moh that have been up for months. This one
crashes every couple of days - the verbose output leading to a crash is
below. Is it just my imagination or has mpg123 always been a pain in the
ass...
What are other mp3 parameters are users using to create mp3s?
John
-- Stopped music on hold o
e like the other box... crapping on 'Ouch ...
error while writing audio data: : Broken pipe'.
John
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Greg,
Without STUN how are the phones able to register? I was unable to get the
Grandstream phones to work at all without STUN.
-John
>From : Greg Oliver <[EMAIL PROTECTED]>
To : Asterisk Users Mailing List - Non-Commercial D
bruary 02, 2008 2:15 PM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Multiple SIP phones behind a Linksys
firewall
>
>
>
> On Feb 2, 2008, at 2:11 PM, John Von Essen <[EMAIL PROTECTED]> wrote:
>
> > I posted an e
Dear List:
Can anyone refer me to a resource to better understand how the SIP protocol
is used by carriers to provide voice circuits between * and the PSTN?
Thanks a bunch!
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there other
open source projects providing these SIP services?
Thanks a bunch!
John
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On Feb 13, 2008 12:33 PM, Tilghman Lesher
<[EMAIL PROTECTED]> wrote:
>
> In the same way that a PHP programmer should not attempt write Python the
> way she writes PHP, I would agree with you. However, if you're willing to
> adapt to the ways the dialplan works, you can create dialplans which aren
I will see what I can find I just joined the list today.
--John
-Original Message-
From: Mojo with Horan & Company, LLC [mailto:[EMAIL PROTECTED]
Sent: Thu 4/3/2008 7:17 PM
To: Asterisk Users Mailing List - Non-Commercial Discussion
Subject: Re: [asterisk-users] Sending audio
You should also get your t1 carrier to provide echo cans on the
circuit. Fairly common and easy for them to set up.
John Chastain
On Sep 4, 2009, at 3:07 PM, Steve Totaro
wrote:
2009/9/4 Vinícius Fontes
No it's not a fact of life. VoIP works as fine as conventional
tele
from my PBX when 220 is ringing or in use. Any help
much appreciated as this has been driving me mad for the last 2 days!
Is this an asterisk config prob (Asterisk 1.4.21.2)?
John
Console output:
-- Executing [...@default:1] SIPAddHeader("SIP/221-08ddaf00",
""Alert-Info:&l
Yes- followed all 3 wiki instructions. Thanks for naming tips! Does
this log help at all? Looks like the PBX isn't sending the SIP
messages- I notice the previous NOTIFY messages said (queued)- does
this mean anything?
John
PBX*CLI> sip show subscriptions
Peer User
tted router on a private IP, the PBX is on a
public IP; could this private IP in the subscriptions be the problem?
John
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New to Asterisk? Join us for a l
ning "Status" and "Addr->IP", can
> 223 be called?
222, 223 + all other extensions say OK (since qualify=yes on), and IP
is the public WAN IP of our router. All extensions can call each other
OK
Thanks for help so far
John
--
__
Have just tried the asterisk setup on an old server that I placed on
our LAN (i.e. server and extensions all on same subnet). BLF worked as
expected, no "queued" messages. Could NAT be the problem?
John
On 6 March 2010 10:16, John wrote:
> added canreinvite=yes
>
>> Do a
You absolute beauty!
SIP ALG was enabled in our router, so that was obviously intercepting
the NOTIFY messages. I've disabled that and the busy lamp works fine,
and I'm getting no (queued) messages
Thanks so much for all your time and help
John
On 8 March 2010 05:09, Philipp vo
it's a Billion BiPAC 7402GXL 3G adsl modem/router running 5.53.s5
John
On 9 March 2010 01:16, Philipp von Klitzing
wrote:
> Hi John!
>
>> You absolute beauty!
>>
>> SIP ALG was enabled in our router, so that was obviously intercepting
>> the NOTIFY messages. I
ry using a hardware phone, or a different softphone e.g. xlite/ linphone
- google asterisk jitter buffer
etc. etc. etc.!
John
On 30 March 2010 15:11, jonas kellens wrote:
> Hello list,
>
> I have set the tos-settings in sip.conf as recommended at
> http://www.voip-info.org/wiki
icaly just a hand off.
John Dunham
John Dunham
CTO - Global Technologies & Communications
713-559-9002
713-559-0001 FAX
(234) 803-6732195 Nigeria GSM
832-922-9123 Cell
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>
John Dunham
can voicemail be setup to allow a calling user into voicemail to access the
the direcotry() ?
or can a voicmail subscriber be setup to send(or forward) a voice mail to
other users using the same directory() feature?
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Try the Wiki, thats where I found it.
John Dunham
Original Message
From: [EMAIL PROTECTED]
To: asterisk-users@lists.digium.com,
Subject: RE: [Asterisk-Users] Asterisk + Radius
Date: Fri, 21 Jan 2005 10:24:20 -0500 (EST)
>Where can i get information about Asterisk using Rad
p me anyway.
Any help or advice on either issue (could they be related?) would be
greatly appreciated.
Regards,
John C.
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27;m an amateur programmer at best, with little experience, not to mention
I'm still learning about the SIP protocol, so I'm just not sure what this
is telling me.
I strongly suspect that this is probably tied in with my situation, but
according to all the docs/forums/setups I've resea
our business clients.
We currently offer many services such as pc2phone,
callback, prepaid calling cards and IP PHONE desktop
solution. From a few meetings and conversations with
our clients we found DID (Direct Inward Dialing) may
be a great service to offer.
Contact me offlist.
John
ay need to make some adjustments
depending on your setup, CID, etc. I used the first one, dropping the CID
portion. Works like a charm.
http://nerdvittles.com/index.php?p=65
http://www.geekgazette.com/index.php?option=com_content&task=view&id=28&Itemid=0&a
I forgot about sugar, which I use, but not enough to have
remembered to back it up, but that's another issue.
Hope this helps answers your question.
Regards,
John C.
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k with both Asterisk and allow passthrough that will not only
transparent, but be less expensive than setting up a UPS system that will
hold the server up for an hour or so. A UPS to hold up the adapting device
and phone for an extended period would be far cheaper, I think.
TIA for
I have a couple of
those laying around in the bottom drawers in just about every room in the
house.
Thanks for a simple answer to what is probably percieved as a "simple"
question. :-)
John C.
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]
>
may be a lousy quick explanation, but I'm
still somewhat of a newb with all this.
Regards,
John C.
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will
> register and 1 that doesn't. This may be a lousy quick explanation, but I'm
> still somewhat of a newb with all this.
>
> Regards,
>
> John C.
>
>
>
> ___
> --Bandwid
With pen in hand, Dean @ INKnBITs succussfully stormed bulwarks which others
armed with sword and excommunication have been repulsed, and said ...
> I have setup a SPA-3000 to forward all incoming PSTN calls to the asterisk
> and for asterisk to use the SPA for outbound calls. This works fine, but
> Does anyone know how many simultaneous calls can a WRTG54GS handle?
> Assuming SIP phones are connected locally using G711.u codec and the
> WRTG54GS connects to a remote Asterisk server using IAX2 trunking
> using GSM codec.
>
> Thanks,
> Daniel
You'll have to do a little experimenting, althoug
the config. Does anyone know a way to do this?
I'm guessing it could be within the file which I have no
idea of the format of. Has anyone got an example or fields to put in
this? I believe you can use it to customize softkeys and stuff so would be
quite useful to see.
On Thu, 23 Mar 2006, john wrote:
Hi,
Does anyone know how to define speeddials in XML for the 7970 sip firmware?.
I've played with the SEP.cnf.xml file that was posted previously but
can't find a way to do it. I can define them on the phone usually (seems a
bit buggy) but if
nd
in other contexts the "#" key should be used to signify the end of a
recording, but pressing that key activates the transfer.
By the way, the attended transfer does not work at all
Any ideas are more than welcomed.
Thanks for the help
John
_
Hi. Thanks a lot for the reply. Script seems good, but i am stuck on how
to make it.
On 27/8/2016 10:48 μμ, Steve Edwards wrote:
On Sat, 27 Aug 2016, tux john wrote:
Hi. I would like to blacklist a few callers and I have been using the
*CLI> database put blacklist 1234 "annoying
Thanks for the reply. i am looking for a guide to help me setup from
scratch this. the OS is raspbian (debian in raspberry). any ideas?
On 12/11/2016 12:55 πμ, Victor Villarreal wrote:
Hi John!
I'm not sure why are you using iaxmodem... I use it a few years ago
with Asterisk 1.4
system for:
fax show version
Cheers,
Larry.
On 15/11/2016 8:09 PM, tux john wrote:
Hi. Since I am messing a lot with it without seeing the end of, may I
ask if there is any solid guide for that please?
On 13/11/2016, 07:42 Larry Moore wrote:
Some additional information which may help yo
I'm looking to get Allison Smith (http://www.theivrvoice.com) to
record a bunch of prompts for me. I sat down and put together a
number of phrases and words that I would expect to be strung together
in a "normal" phone system. I have not been able to find a set of
"standard" words that one wo
noise is low on the echo_can priority list until
the basic functionality becomes more ideal.
John
>>We can all agree that the Aggressive Suppression code has gone a long way
to solving the echo problems of Asterisk. I, and others, have noticed that
its methods do seem a little harsh at time b
Where can I buy and for how much?
John
-
NetRom Internet Services973-208-1339 voice
[EMAIL PROTECTED] 973-208-0942 fax
http://www.netrom.com
I know I heard people looking for inexpensive ADSI phones a while
back. Can't vouch for these guys, but this looks like a reasonable
deal for what perhaps are new phones. No experience with ADSI,
myself, but thought I'd pass it along. These folks also carry the
Nortel analog units that do in
ne of the jobs has used a significant
timeslice - so the others might be zombies.
John Harragin
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>These are threads, the 7 megs each of them shows is the total for the
whole process, not 7 meg per child. They're safe and a normal side
effect of the design.
Thanks, must have changed the default settings of ps.
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I get these errors (480 "Temporarily...") when I try to use my
iconnect account quickly after hanging up on a previous session.
They have some sort of contention locking system which allows only
one call at a time on an account, and if you do not give it adequate
time to "settle", you'll hit t
to an abnormally high sensitivity to static-electric
shock. I can only assume that it got zapped once two many times and now
no longer works.
I'm not really sure what to do now. We can't afford a channel bank and
it hardly seems sensible to purchase another S100U.
Any suggestions?
--
Joh
Just a note to the list, I tried to apply that patch to the current CVS
but it failed. This was expected of course because the code has changed
since the patch was released.
Assuming this patch is stable it should defiantly be Incorporated into
the main code.
John Lange
On Tue, 2003-03-04 at 23
I have recently begun experimenting with Asterisk, and have been
mightily impressed by its capabilities and flexibility. I have run
across one problem, however, that challenges my ability to use it as a
production system.
My Asterisk box has a public Internet IP, and works great with SIP
Finally someone has hit the same problems that we have. Everyone on this
newsgroup seems to have static IPs!
The problems you get can manifest in 2 ways:
1) you cannot get through to the phone at all
2) one-way audio - you can hear the other end but they can't hear you.
The problem is a combinat
Symptoms: when calling my iconnect phone number (13033913323 in my
bogus example below) from my cell phone, I can see that the call
makes it to my asterisk server, and my phones even ring once as *
passes the call through during the "180 Ringing" period. However, it
seems that iconnecthere.com
Try in sip.config
disallow=all
allow=ulaw
Select 30ms packet size in the snom.
John
- Original Message -
From: "David Davis" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Thursday, March 06, 2003 2:40 PM
Subject: RE: [Asterisk-Users] SNOM sound quality
>
Thanks, Mark!
Here's a summary of what one needs to do in order to get NAT working
with Asterisk. Please note that I have a Cisco ATA-186, and your
experience may be slightly different based on the equipment you're
using. You'll need to have a CVS updated version of Asterisk as
2003-03-06 ~2
I don't think it is a filter (SIP uses UDP, so telneting to the port
(which uses TCP) doesn't mean much.
There is a problem right now with receiving incoming calls from
iconnecthere/d3. Something on d3's end is messed up. It hits the
asterisk server, and then cancels the connection. I'm stumped.
This may be slight off topic, but perhaps it has relevance:
My iconnecthere account no longer works for "inbound" calls through
NAT using the standard configuration that they provide on their
website. I have sent them a message, but I believe it will be
flushed down the toilet by the first-tie
ED] [mailto:asterisk-users-
[EMAIL PROTECTED] On Behalf Of Roderick Montgomery
Sent: Thursday, March 06, 2003 6:40 PM
To: [EMAIL PROTECTED]
Subject: Re: [Asterisk-Users] NAT working outbound with Asterisk and ATA-
186 phones
Thanks to Mark and John!
With Mark's "nat=1" changes y
> * iax (I
have wondered if there is an iax directive analogous to packet size), but
also sounds good with ulaw.
John
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Just out of curiosity, has anyone been working on adding ogg or mp3 encoding
to voicemail? Lame sounds quite good at low bit rates. Anyway, either of
these would probably be good for the email message distribution.
John
>> It appears that the pharsing for the wav49 extension which is .WAV
I didn't mean mp3 or ogg for playback, but emailed voicemail files. The
original wav paths could be could be dumped into a low priority encoding
queue (for that matter any additional conversion could be done in this
fashion).
By the way do all holded channels hear the same signal?
To: [EMAIL PROTECTED]
From: John Todd <[EMAIL PROTECTED]>
Subject: NAT working outbound with Asterisk and ATA-186 phones
Thanks, Mark!
Here's a summary of what one needs to do in order to get NAT working
with Asterisk. Please note that I have a Cisco ATA-186, and your
experience may
Hmm... haven't been able to get this to work on my Cisco ATA-186.
Perhaps I'm trying the incorrect knobs? I'm making outbound calls
ATA-186->*->iconnecthere->PSTN.
I've set my ATA-186 to these various settings:
AudioMode: 0x00150015
AudioMode: 0x00250025
AudioMode: 0x00050005
(per the settings
Is there a way to configure a channel to ethernet, such that we
could use one of the VoIP long distance carriers, like Vonage?
Yes.
See http://www.loligo.com/asterisk/
for some sample configs using iconnect.
vonage does not (supposedly) allow people to use the
username/password they give yo
My example had a Macro to do the dialing towards iconnect, since I
got really tired of typing in the same dial routines (obviousl,
that's what Macros are intended to replace.)
So let's simplify my example and your configs. You have some extra
stuff in there that probably you should not have on
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.
That should have no
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
xten => s,3,DigitTimeout,10
Move up or eliminate wait
;>> exten => s,2,Wait,1
>> exten => s,1,Answer
>> exten => s,3,DigitTimeout,10
John Harragin
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rt of resembles a double press. I have heard this with the different
dtmfmodes.
John
- Original Message -
From: "Mark Spencer" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Wednesday, March 12, 2003 6:51 PM
Subject: Re: [Asterisk-Users] iconnecthere DTMF solution?
I'm in favor of "TASTE" myself, though Mark's take on "LIVE" has the
all-important "I", to establish the use of this protocol over IP
networks, which is an important part of the protocol and conceptual
structure, yes?
Perhaps "ITASTE" with the "I" standing for the obvious "Internet".
JT
Wha
o iax stream include version information. Did this wind up in the
specification?
John
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ormat
32 equal ADPCM but what are the others?
TIA
John
-
NetRom Internet Services973-208-1339 voice
[EMAIL PROTECTED] 973-208-0942 fax
h
Thanks for the feedback but I'm still lost on this one (Forgive my
ignorance please)
I don't understand how "#define AST_FORMAT_ADPCM(1 << 5)" becomes
a "format = 32" in t
326]: File chan_iax2.c, Line 2999 (authenticate): No way to send
secret to peer 'xxx.xxx.xxx.xxx' (their methods: 4)
Everything still works fine but should I be concerned about this or is my
iax.conf missing some
Why would you use anything other than what's in the sip.conf file?
You can now configure (though I have not tried it myself) usernames
with "@" symbols in them. I am having a lot of difficulty parsing
your question. Just give people usernames of
"[EMAIL PROTECTED]" then and map them to some
s long as T1s have been around, it seems
obvious that there should be codecs especially for this purpose. Does anyone
know if this is the case?
John
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I believe the Cisco ATA-186 supports it, but you'd have to do more
digging on their site.
This is really not a protocol issue, but a vendor programming issue.
It all depends on if you can get the hardware to do a hotline call
when the phone is taken off the hook.
JT
Does anyone know if this
This (fraud?) was brought to my attention today. An interesting
idea, I don't know if it's genuine or not, as I have not purchased
it. What interests me more is the concept behind the product: the
ability to generate CID tones after-the-fact in order to populate
caller ID devices with customi
Just an FYI but I'm seeing the same thing using ata-186's
John
-
NetRom Internet Services973-208-1339 voice
[EMAIL PROTECTED] 973-208-094
ESSIVE_SUPPRESSOR in the zaptel
Makefile then make install.
John
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http://www.xiph.org/press/2003/nonprofitspeex1/
John
-
NetRom Internet Services973-208-1339 voice
[EMAIL PROTECTED] 973-208-0942 fax
http://www.netrom.com
ating issues into documentation.
extension.conf
exten => _1114,1,Dial,SIP/sip:1114|5
sip.conf
[sip:1114]
type=friend
user=john
context=default
secret=pword
dtmfmode=inband
mailbox=1114
host=192.168.0.114
On the sip/snom200/Authentication page:
Realm=
username=john
password=pword
On the sip/snom2
vacy.o] Error 1
John
-
NetRom Internet Services973-208-1339 voice
[EMAIL PROTECTED] 973-208-0942 fax
http:/
I have my snom200 registering only if I set the passwords to blank in both *
and the snom...
sip.conf
secret=
SNOM/Home/Settings/SIP/Authentication
Realm=
username=1114
password=
...I'm wondering if there is a setting in the snom that requires an
encripted password?
Thanks,
John
This e
On Fri, 28 Mar 2003, Stephen Davies wrote:
>
>
> On Fri, 28 Mar 2003, Mark Spencer wrote:
>
> > Last night I committed SIP retransmission support into Asterisk. Let me
> > know if this helps/hurts for anyone. Thanks!
>
> You set a 15 second autokill timer but never cancel it. So calls tend to
>
to mute the speakers and engage a mic. With this as
default intercom mode it could eliminate echo issues and prevent snooping.
John
(Sorry if this appears twice as I accidentally posted from the wrong
address)
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> Do not plug in ethernet cable at all, even if the interface is down.
This device has poor networking implementation. If I leave one of ours
plugged in (pretty traffic intensive in our shop) it is sure to go down. If
it is in a switch or slow segement it does ok but as Steven pointed out you
are
ittle coat
of dust. I should probably pop it open and vacuum it out.
John
> The power supply turned out to be fairly inexpensive at $35US, so I wasn't
too
ticked off. :) If you or anyone else needs a source for the power supply,
let me know.
-wade
This e-mail was scanned and found cl
used ports and I see
messages something like "Picking up channel Zap/9-1" then a few seconds
later Hanging up...
John
- Original Message -
From: "Gene Kochanowsky" <[EMAIL PROTECTED]>
To: <[EMAIL PROTECTED]>
Sent: Tuesday, April 01, 2003 6:13 PM
Subjec
Hey do we have the ability to incriment a variable?
exten => t,2,SetVar,looptest=$((looptest + 1))
I was thinking of doing a library of simple arithmetic and bash-like
expansions for asterisk... like Zap/{1&,2&,3} - but it may already have this
functionality.
John
> exten
?
John
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>disable echo cancellation
Can we currently do that on a per-channel basis? For our situation. * to iax
to * to pbx, I need it on - as most of the traffic is voice.
John
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Brian Jones
Sent: Thursday, April
James,
> I have to disagree here. I send and receive faxes over IAX all the time
What echo_cans are you using on each end and do you have daggressive
suppression enabled (one or both ends)?
John
This e-mail was scanned and found clean by Monroe-Woodbury CSD Antivi
tems. The
cable has a amphenol male D50 connector on one end and probably a rj45 on
the other. I also don't know if any other devices are needed between the
System75 and asterisk The DS1 Tie Trunk card is a TN722B.
Any information would be helpful.
Thanks,
John
This e-mail was scanned and
ormal RJ45 do we transmit (Out) on 1&2 or 4&5?
John
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of John Harragin
Sent: Friday, April 04, 2003 2:25 PM
To: Asterisk
Subject: [Asterisk-Users] AT&T T1 Cable Needed!
Hi,
I just got a T1 interface f
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